Re: [asterisk-users] The purpose of DUNDi
dave cantera wrote: remco, et al, could I use dundi where I could use an area code to determine the connecting server or dial string? just like we would use 88XXX to dial a 3 digit extension on another server at location 88? or dial 84XXX for a 3 digit extension on a server located at 84?... yes you can. You'll setup a context in your dialplan on your server where you'll tell dundi that you accept calls for say _88XXX and have a mapping for that context in your dundi.conf thanks, daveC Remco Post wrote: Rilawich Ango wrote: It is quite interesting and I am looking for it. Could you give me some more information or website how to set it up? Have a look at: http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf and the two links at: http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_exec: Unable to join queue
lenz wrote: Is the queue enidan configured at all in queues.conf? and how is it defined? l. Sorry, I should have added that: from queues.conf: [enidan] strategy = ringall ;announce = enidan-queue member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] Also, what I discovered yesterday is the following: just after an asterisk restart: *CLI show queue enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet Local/[EMAIL PROTECTED] (Invalid) has taken no calls yet No Callers The (Invalid) bit is worrying, but after a reload of app_queue: *CLI show queue enidan has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (Not in use) has taken no calls yet No Callers /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
On Tue, 15 May 2007, Vincent Delporte wrote: Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). = When a call comes in, I'd like an AGI application to send an e-mail and send CID name/number to a script on a web server. Is this the correct way to do it in Perl, with the modules available in AsteriskWin32? Could I rewrite this in Delphi instead? ALL AGI scripts are basically just programs that read from stdin and write to stdout. They can therefore be written in almost any language. So yes, Delphi should work fine. (I have very fond memories of Delphi, and before that, Borland Pascal w/ Objects for DOS, and before that, Turbo Pascal... one of these days I'll have to get the latest version of Delphi and take a walk down memory lane. These days everything is C this or Perl that. I loved Pascal. :-)) -- Nick ___ --Bandwidth 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] Req-Installation process for app_dtmftotext.c
Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c , And I already tried with 'spandsp' application for this. But I am getting errors.[I followed the instructions at http://www.soft-switch.org/installing-spandsp.html] specifically by running this command:patch apps_makefile.patch I need clarification on 'ld.so.conf' file.[It has to be in the /etc/ directory. If you do not have such file - make one. In the file you need to add the path to the spandsp library.] Please give me the steps for this step. I installed asterisk 1.2.17 only, i not installed any libpri or zaptel sources. Can anybody be of help Me on this getting DTMFToText() application on asterisk with the help of app_dtmftotext.c and/or spandsp application is appreciated. Regards K.Rajesh. _ Spice up your IM conversations. New, colorful and animated emoticons. Get chatting! http://server1.msn.co.in/SP05/emoticons/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [*Win32 0.60] Sending call notification bye-mail/web?
I'm using an FastAGI written in Delphi for my IVR so I can confirm it works just fine. I wrote all the code from scratch and it wasn't a big deal, but you can find sample code on Free Pascal sites (google will help you). Also I'd recommend turning your idea into an FastAGI. It will work with both native (Linux) Asterisk and with the Win32 port, and it will actually be easier to debug! You just start your FastAGI server exe, place a brakepoint in the code, pick up your phone and dial your test number. Asterisk has long-enough timeouts when talking to an FastAGI application to make stepping through the code possible. -- Cosmin Prund -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Seraphin Sent: Tuesday, May 15, 2007 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [*Win32 0.60] Sending call notification bye-mail/web? On Tue, 15 May 2007, Vincent Delporte wrote: Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). = When a call comes in, I'd like an AGI application to send an e-mail and send CID name/number to a script on a web server. Is this the correct way to do it in Perl, with the modules available in AsteriskWin32? Could I rewrite this in Delphi instead? ALL AGI scripts are basically just programs that read from stdin and write to stdout. They can therefore be written in almost any language. So yes, Delphi should work fine. (I have very fond memories of Delphi, and before that, Borland Pascal w/ Objects for DOS, and before that, Turbo Pascal... one of these days I'll have to get the latest version of Delphi and take a walk down memory lane. These days everything is C this or Perl that. I loved Pascal. :-)) -- Nick ___ --Bandwidth 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] Some problems with mysql CDR
Hello,
Is your userfield type varchar(255)?
Also check if you edited the cdr_addon_mysql.c and Make file to tell
cdr_addon_mysql.c to store uniqueid as outlined here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
This happens when you have UNIQUE for some field and insert same record twice.
In order to help please paste your [cdr] table structure.
I'm sure it's not a bug but misconfiguration which can be solved easily.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP ServicesSolutions
MOR - FREE Open Source billing for Asterisk
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Martin
Sent: Monday, May 14, 2007 10:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Some problems with mysql CDR
Hello,
We have finally upgraded to Asterisk 1.4, however we've run into two issues
that weren't occurring before the upgrade.
Issue #1: We're an outgoing call center and need to record all calls. We use
the uniqueid field in the CDR to match with the recording, which we labeled
with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct
in the CDR. Here is the manager event for a call:
Event: Cdr
Privilege: call,all
AccountCode: 6384106:MMI-Y:200705081051010077
Source: 00
Destination: 6398714109927773
DestinationContext: outbound
CallerID: 00
Channel: Zap/15-1
DestinationChannel: SIP/teliax-081ed5b0
LastApplication: NoOp
LastData:
StartTime: 2007-05-08 10:51:04
AnswerTime: 2007-05-08 10:51:05
EndTime: 2007-05-08 11:01:56
Duration: 652
BillableSeconds: 651
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: 1178635864.1510
UserField:
And for that record in the database:
'calldate' '2007-05-08 10:51:04'
'clid' '00'
'src' '00'
'dst' '6398714109927773'
'dcontext' 'outbound'
'channel' 'Zap/15-1'
'dstchannel' 'SIP/teliax-081ed5b0'
'lastapp' 'NoOp'
'lastdata' '',
'duration' 652,
'billsec' 651,
'disposition' 'ANSWERED',
'amaflags' 3,
'accountcode' '6384106:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn't answered,
Asterisk complains:
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
I haven't found any other information regarding these errors. I am just
wondering if they are bugs. Any insight would be appreciated!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
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Re: [asterisk-users] force outgoinc callerid
On 5/10/07, Steve Kennedy [EMAIL PROTECTED] wrote:
On Thu, May 10, 2007 at 06:36:53PM +0200, nik600 wrote:
i have a Te205P connected to a PRI E1, can i force the outgoing
callerid to change for each context?
for example:
[outgoing_context_one]
;force callerid to 12345
exten = _XXX,1,Set(CALLERID(number)=12345)
exten = _XXX,1,Dial(Zap/${EXTEN})
[outgoing_context_two]
;force callerid to 2
exten = _XXX,1,Set(CALLERID(number)=2)
exten = _XXX,1,Dial(Zap/${EXTEN})
Can i do that?
thanks to all
Assuming your telco allows it.
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
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Hi i've checked and my telco allows to pass the callerid.
But i still have problems with the configuration, i've enabled the
logging of pri and this is the output:
-- Making new call for cr 32797
-- Requested transfer capability: 0x00 - SPEECH
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 2 (reference 29/0x1D) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 96]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 22 ]
[6c 02 00 c3]
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown
Number Plan (0)
Presentation: Number not available (67) '' ]
[333xx]
Called Number (len=13) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '333x' ]
Note:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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[asterisk-users] Re: Difference between making a call and Originate
Christopher Robinson wrote: When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=SIP/[EMAIL PROTECTED], 'Context'='mycontext', 'Exten'='899', 'Priority'=1, 'Callerid'='whatever')); It creates a screech sound when the first audio file is played. Doesn't seem to happen with another VSP I tried, but still, why would a regular outbound call work just fine and Originate create this strange sound. I know for sure that it isn't the audio file that I'm playing by the way. I too have noticed this. I'm taking a stab in the dark here but is it possibly voice packets of a different codec being decoded as garbage/static? Our issue mysteriously went away. Don't know exactly what caused it though unfortunately. ___ --Bandwidth 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] cpu usuage
Do any one knows the formula to calculate memory and cpu usuage for channel on g729 codec,to know the hardware required for 100 concurrent call. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cpu usuage
Does such formula exist ? And do you have other functions/apps that demands cpu power that needs to be taken into the formula. And please skip that disclaimer you have in the bottom ! /Mats On 5/15/07, Khaled Chehab [EMAIL PROTECTED] wrote: Do any one knows the formula to calculate memory and cpu usuage for channel on g729 codec,to know the hardware required for 100 concurrent call. 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] HPEC audio clipping
If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. Note, however, that we have observed stability issues with HPEC 8.2 (two kernel panics in two weeks since installing HPEC). (The 9.00 version had such severe clipping that we could not run it long enough to determine if it was more stable). g. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Proper AGI use with MySQL
In article [EMAIL PROTECTED], [EMAIL PROTECTED] wrote: All this seems to be working. However, we just don't feel we are doing things properly and reading up on the wiki more about AGI and dialing out, etc, just makes me feel we could be doing things better. Here are some of the things we think we could be doing better but are not sure: 1) Ideally, we would like for the AGI script to know when the call hangs up so that it properly updates callend without having to run the DeadAGI command in the h extension. IMHO, using DeadAGI in the 'h' extension is the correct way to do it. I use that technique in many applications and have found it reliable. I'm not sure why you feel it is not the proper way. 2) We would like for the AGI script to stay running for the life of the call and keep in memory all the user's IVR selections until the call is hung up. At which point, we could actually INSERT the row in MySQL with all the data, instead of constantly hitting the database with updates. You then have the (hopefully rare) problem that if the script, or asterisk, or the box goes down before you do the final insert, you have lost all the selections. I would do the updates as they happen. MySQL is pretty good about caching records, so multiple updates to the same record should get optimised quite well. 3) We read on the wiki the following: If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. In our IVR, we always exit with -1. So, this statement confused us. Does it mean that when we transfer the call to the queue, we should actually return 0 instead of -1 to indicate that the AGI is still running? Can anyone explain this further? Can't comment on this one, as I never use AGI to dial. My AGIs just set the context, extension and priority, and exit to the dialplan to do any dialling. 4) When should we close the database handle? Currently, we have it at the end of the AGI script and also as part of the DeadAGI script. However, which one is actually closing it, we don't know. An AGI is a separate process, so will need to create its own database handle by calling mysql_connect(). For tidiness, it should release the handle by calling mysql_close() before exiting, but if it doesn't, the handle will be closed automatically as the process exits. There is no way the DeadAGI script could inherit a database handle from the original AGI acript. Hope this helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapateller and IAX2#
On Sun, May 13, 2007 at 01:00:27PM +0100, --[ UxBoD ]-- wrote: Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come across this ? I am using * 1.4.4. If you are calling Zapateller without options, you WILL get the Special Information Tone - that's what it's for. The option nocallerid will prevent it playing if caller ID is presented. If you are saying you are not getting a ringing tone, that is perhaps unsurprising. I get no ringing tone on automatically answered IAX2 numbers either. So, what is your intention in using Zapateller? -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapateller and IAX2#
Hmmm, I do have answer() in the dialplan. When I was using with the BRI I did get the single SIT tone. My intention for Zapateller is to play the SIT tone to try and stop autodiallers. On Tue, 15 May 2007 12:00:35 +0100, Phil Reynolds [EMAIL PROTECTED] wrote: On Sun, May 13, 2007 at 01:00:27PM +0100, --[ UxBoD ]-- wrote: Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come across this ? I am using * 1.4.4. If you are calling Zapateller without options, you WILL get the Special Information Tone - that's what it's for. The option nocallerid will prevent it playing if caller ID is presented. If you are saying you are not getting a ringing tone, that is perhaps unsurprising. I get no ringing tone on automatically answered IAX2 numbers either. So, what is your intention in using Zapateller? -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 (0) 845 869 2749 SIP: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapateller and IAX2#
On Tue, May 15, 2007 at 12:23:56PM +0100, --[ UxBoD ]-- wrote: Hmmm, I do have answer() in the dialplan. When I was using with the BRI I did get the single SIT tone. My intention for Zapateller is to play the SIT tone to try and stop autodiallers. Single SIT tone? Do you mean you now get it twice? Or was it in fact being trimmed to just the last tone? I find it's worth pausing slightly before using Zapateller. I use Zapateller before my no caller ID, void caller ID, and redlist messages - and I get the three tones (which is the SIT). I always Answer and Wait(2) before it. Most autodialler systems will tend not to present a caller ID, and those that do quickly get blacklisted. Grossly abusive ones may find themselves redlisted. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth 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] Originate and ForkCDR()
Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the answer field is always empty, billsec field is 0 and disposition field is always set to NO ANSWER. Is there something I'm missing? Thanks, Federico -- Federico Cabiddu RD Software Engineering Abbeynet S.p.A. - www.abbeynet.com http://www.abbeynet.com phone: +39 070 2339349 http://www.federico_cabiddu.sitofono.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
Could I rewrite this in Delphi instead? I never used Delphi to write an AGI but I've seen a class in FreePascal that you could probably use as a base : http://www.automated.it/asterisk/fpc-agi.html hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to set Name/username to something like 229/john instead of 229/229
Hi, 'sip show peers' display info this way Name/username HostDyn Nat ACL Port Status 229/229xxx.39.12.58 D N 63969OK (26 ms) 228/228xxx.39.12.58 D N 63961OK (32 ms) But this makes is difficult to understand which extension belongs to whom, and I want to set it so I can see it like this: Name/username HostDyn Nat ACL Port Status 229/johnxxx.39.12.58 D N 63969OK (26 ms) 228/mikexxx.39.12.58 D N 63961OK (32 ms) In case of Grandstream GXP-2000 phones, I've setup in sip.conf extensions declaration like this: [229] username=john fromuser=229 type=friend secret=229 host=dynamic dtmfmode=rfc2833 dial=SIP/229 context=mycontext And in Grandstream account setting: SIP User ID: john Authenticate ID: 229 But it doesn't work and I see the following Name/username HostDyn Nat ACL Port Status 229/john(Unspecified)D N 0UNKNOWN I've tried to play around with the settings, but no success and the phone doesn't get registered. Anybody who has done this before successfully please help. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] `PATH_MAX' undeclared here (not in a function) in asterisk!
On 5/15/07, lizhong zhu [EMAIL PROTECTED] wrote: I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run ./configure and menuselect with embedded modules. but running make comes out errors: ranlib libmxml.a [...[ /usr/src/asterisk-1.4.2/include/asterisk/paths.h:23: `PATH_MAX' undeclared here (not in a function) [...] PATH_MAX on the Linux systems I have comes from /usr/include/linux/limits.h, which gets pulled in by a few headers, sys/param.h being the most used one. In my 1.4.4 source tree, this gets pulled in via autoconf, which has this snippet in it's output file include/asterisk/autoconfig.h /* Define to 1 if you have the sys/param.h header file. */ #define HAVE_SYS_PARAM_H 1 Check that you indeed have all your headers installed. If PATH_MAX is in an include file, but not one that gets pulled in by including sys/param.h, then the configure script might need to be updated - best to open a bug and attach your config.log as well as the basic info about your system. I suspect that you're just missing the kernel-headers rpm (or equiv for your Linux flavor). That's where I get my linux/limits.h from: [EMAIL PROTECTED] asterisk-1.4.4]# rpm -q --whatprovides /usr/include/linux/limits.h kernel-headers-2.6.18-8.1.3.el5 [EMAIL PROTECTED] asterisk-1.4.4]# Many default installs do not include the kernel headers - you either have to choose a kernel development package bundle at install time or install them manually after the fact. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Web based call control
It may not be exactly what you are looking for, but agentcallbacklogin with ackcall=yes requires the # key to be pressed to answer a call from a queue. We use this to avoid the possibility that the call ends up in a cellular or home voicemail. You can set the queued call to ring instead of music. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Monday, May 14, 2007 9:45 PM To: asterisk-users@lists.digium.com Subject: Web based call control Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will take to go to their home phone voicemail, but I don't want to deliver the call there, I want it to go to the next priority in asterisk. So I was thinking that it would be nice to build a web interface that they could have a button to answer with. This would send a manager command to the server telling it to answer the channel, any thoughts on how to do this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Capacity
What about using a Lucent TNT and an asterisk box. Is the limit the TDM--G.711 conversion? On 5/14/07, Alex Balashov [EMAIL PROTECTED] wrote: On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? -- 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
[asterisk-users] Mr. Spencer Written
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers http://www.voip-magazine.com/content/view/3644/0/1/0/ in the
VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
I didn't get understand the usage ARG1 argument in the context. Do use
ARG1 variable only into macro? Is not this usage apparent contradiction
with Asterisk documents? Do anyone get explain this?
Regards,
Cleviton
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Re: [asterisk-users] Asterisk High-Capacity Stability
On May 14, 2007, at 11:27 PM, Atlanticnynex wrote: I'm curious what kind of configuration/features/modules you could recommend for my setup. Can you explain further what you mean by OpenSER to Asterisk? If you want to go Open Source, I think OpenSER is a good choice. You won't need to do any hacking to make it work..I'd suggest making 1 or 2 openser boxes to act as registrars for your user agents, and use the openser dispatcher module to point at one or more openser boxes that do LCR for calls that go directly out, and at one or more asterisk boxes for feature servers if you need them. Using Asterisk realtime and the database extensions for OpenSER you can share the user database between them and things should just work. Write your CDRs to a separate database (as to separate business data and call flow datajust in case someone does a complex CDR query you don't want your PDD to go through the roof) and come up with some kind of CDR remediataion for billing. ___ --Bandwidth 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] finding the sipp soft phone list on the wikey
Hello everyone, I am new to Asterisk and I am trying to find the list of sip soft phones list but I am having trouble finding the list. Can some one point me to a url where I could find this? I have tried looking for this myself and found it twice but now I can't find it again. Thanks much. Scott___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom 501 configuration setting
I recently got a polycom 501. I was trying to get the phone to accept the TFTP boot files. I was REALLY confused when I finally figured out that the phone does FTP by default and you have to go change it to TFTP using the keyboard menus to switch it to TFTP. Am I missing something here? I certainly would have thought the phone would be intelligent enough to try a FTP first - If you dont get what you want, then try TFTP then HTTP or what ever My question is: Is there a different method (other than using ftp) to have the phone by default use TFTP and grab its config without getting into the keyboard menus? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] finding the sipp soft phone list on the wikey
On 5/15/07, Scott Berry [EMAIL PROTECTED] wrote: I am new to Asterisk and I am trying to find the list of sip soft phones list but I am having trouble finding the list. Can some one point me to a url where I could find this? I have tried looking for this myself and found it twice but now I can't find it again. Thanks much. Use the search. The page you're looking for will return as the first link if you search for softphones. -erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mr. Spencer Written
On 5/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers in the VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
I didn't get understand the usage ARG1 argument in the context. Do use ARG1
variable only into macro? Is not this usage apparent contradiction with
Asterisk documents? Do anyone get explain this?
Regards,
Cleviton
Cleviton,
You can pass arguments into a macro:
exten = 500,1,Macro(something,${EXTEN})
[macro-something]
exten = s,1,Dial(SIP/${ARG1},60)
This is perfectly valid. You can pass multiple arguments and read
them with ${ARG1}, ${ARG2}, etc, etc:
http://www.voip-info.org/wiki-Asterisk+variables#Macrospecificvariables
--
Kristian Kielhofner
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RE: [asterisk-users] IAX2 peer unreachable in one direction - NATproblem?
To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seb Auriol Sent: 14 May 2007 19:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] IAX2 peer unreachable in one direction - NATproblem? The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below with iax2 debug and core debug 3. I actually have an Asterisk 1.2 and an Asterisk 1.4 server in the non-natted role, and both have the same issue. However, I have another non-natted server (on a different ISP) that can talk fine to the natted server. (IP addresses replaced with names.) myNonNattedServer*CLI iax2 show peers Name/UsernameHost Mask Port Status myNattedServUN myNattedServer (S) 255.255.255.255 4569 (T) UNREACHABLE [May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1154 update_max_nontrunk: New max nontrunk callno is 7 [May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1252 find_callno: Creating new call structure 6 [May 14 19:06:05] DEBUG[5551]: chan_iax2.c:1644 send_packet: Sending 12 on 6/0 to myNattedServer:4569 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00012ms SCall: 6 DCall: 0 [myNattedServer:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00012ms SCall: 5 DCall: 6 [myNattedServer:37657] Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 5 [myNattedServer:37657] [May 14 19:06:05] DEBUG[5546]: chan_iax2.c:4788 raw_hangup: Raw Hangup myNattedServer:37657, src=6, dst=5 [May 14 19:06:06] DEBUG[5540]: chan_iax2.c:1644 send_packet: Sending 12 on 6/0 to myNattedServer:4569 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00012ms SCall: 6 DCall: 0 [myNattedServer:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00012ms SCall: 6 DCall: 6 [myNattedServer:37657] Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 6 [myNattedServer:37657] [May 14 19:06:06] DEBUG[5542]: chan_iax2.c:4788 raw_hangup: Raw Hangup myNattedServer:37657, src=6, dst=6 myNattedServer*CLI iax2 show peers Name/UsernameHost Mask Port Status myNonNattedSeUN myNonNattedServ (S) 255.255.255.255 4569 (T) OK (14 ms) Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00016ms SCall: 00010 DCall: 0 [myNonNattedServ:4569] May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1007 update_max_nontrunk: New max nontrunk callno is 12 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1112 find_callno: Creating new call structure 11 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet 0, (6, 30) May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 30 received May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6857 socket_read: For call=11, set last=16 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1515 send_packet: Sending 16 on 11/10 to myNonNattedServ:4569 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00016ms SCall: 00011 DCall: 00010 [myNonNattedServ:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 00010 DCall: 00011 [myNonNattedServ:4569] May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet 0, (6, 10) May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 10 received May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7510 socket_read: Immediately destroying 11, having received INVAL May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7513 socket_read: Destroying call 11 Also when calls are placed to myNonNattedServer from myNattedServer (which does work), the channel name is IAX2/myNattedServer:37657-callno, as opposed to IAX2/myNattedServUserName-53. (BTW, if I turn off qualify on
Re: [asterisk-users] polycom 501 configuration setting
Hi: Jerry Geis wrote: I recently got a polycom 501. I was trying to get the phone to accept the TFTP boot files. I was REALLY confused when I finally figured out that the phone does FTP by default and you have to go change it to TFTP using the keyboard menus to switch it to TFTP. You only have to change this once; after that the phone will retrieve configs with TFTP first. Many people don't bother with TFTP anymore, not least because it's even less secure than FTP. HTTPS is best for security. (We use TFTP but only because we're on a private network). Am I missing something here? I certainly would have thought the phone would be intelligent enough to try a FTP first - If you dont get what you want, then try TFTP then HTTP or what ever My question is: Is there a different method (other than using ftp) to have the phone by default use TFTP and grab its config without getting into the keyboard menus? Sadly, no. This means that if you have 500 sets to configure for TFTP, you'll have to change the setting manually. I would be happy to be corrected on this. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Mr. Spencer Written
Kristian Kielhofner wrote:
On 5/15/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of
Asterisk Servers in the VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
I didn't get understand the usage ARG1 argument in the context. Do
use ARG1 variable only into macro? Is not this usage apparent
contradiction with Asterisk documents? Do anyone get explain this?
Regards,
Cleviton
Cleviton,
You can pass arguments into a macro:
exten = 500,1,Macro(something,${EXTEN})
[macro-something]
exten = s,1,Dial(SIP/${ARG1},60)
This is perfectly valid. You can pass multiple arguments and read
them with ${ARG1}, ${ARG2}, etc, etc:
http://www.voip-info.org/wiki-Asterisk+variables#Macrospecific
variables
Kristian, I think Cleviton's point was that [lookupdundi] does not appear to
be a macro, and so how can it have arguments if it just a normal context /
extension?
Kind regards,
Sebastian
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[asterisk-users] Trixbox problems
Hello, I'm writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) - Sometimes when the user dials out, he hears the ringing tone but the line is already answered and the called party hears his voice while he's still hearing the ringing tone. How can I investigate those 2 problems in order to find what's happening ? Thanks marco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Originate and ForkCDR()
Hello, I guess you have something like this: ACTION: Originate Channel: Local/1234 Exten: 4321 Priority: 1 Context: blabla And in [blabla] Exten = 4321,1,Dial(something. Instead use magic /n setting with Local channel. See here: http://www.voip-info.org/wiki/view/Asterisk+local+channels And do your originate like this: ACTION: Originate Channel: Local/[EMAIL PROTECTED]/n Exten: 4321 Priority: 1 Context: blabla And in [blabla] Exten = 4321,1,Dial(Local/something/n) I guess you got the idea. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP SolutionsServices MOR - FREE Open Source billing for Asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Cabiddu Sent: Tuesday, May 15, 2007 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Originate and ForkCDR() Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the answer field is always empty, billsec field is 0 and disposition field is always set to NO ANSWER. Is there something I'm missing? Thanks, Federico -- Federico Cabiddu RD Software Engineering Abbeynet S.p.A. - www.abbeynet.com http://www.abbeynet.com phone: +39 070 2339349 http://www.federico_cabiddu.sitofono.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [asterisk-users] Mr. Spencer Written
Perfect, Auriol!
Exactly this.
Cléviton.
-Mensagem original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Seb Auriol
Enviada em: terça-feira, 15 de maio de 2007 12:45
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Assunto: RE: [asterisk-users] Mr. Spencer Written
Kristian Kielhofner wrote:
On 5/15/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Mr. Spencer written the article Using DUNDi with a Cluster of
Asterisk Servers in the VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
I didn't get understand the usage ARG1 argument in the context. Do
use ARG1 variable only into macro? Is not this usage apparent
contradiction with Asterisk documents? Do anyone get explain this?
Regards,
Cleviton
Cleviton,
You can pass arguments into a macro:
exten = 500,1,Macro(something,${EXTEN})
[macro-something]
exten = s,1,Dial(SIP/${ARG1},60)
This is perfectly valid. You can pass multiple arguments and read
them with ${ARG1}, ${ARG2}, etc, etc:
http://www.voip-info.org/wiki-Asterisk+variables#Macrospecificvariables
Kristian, I think Cleviton's point was that [lookupdundi] does not appear to
be a macro, and so how can it have arguments if it just a normal context /
extension?
Kind regards,
Sebastian
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Re: [asterisk-users] Trixbox problems
On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote: Hello, I’m writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) - Sometimes when the user dials out, he hears the ringing tone but the line is already answered and the called party hears his voice while he’s still hearing the ringing tone. How can I investigate those 2 problems in order to find what’s happening ? Contact the Trixbox mailing lists? -- Dave Cotton [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
[asterisk-users] Outside lines are just not happening...
Two problems, possibly related:
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
exten = _9NXX,1,Dial(Zap/5/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
I don't know if it's relevant or not, but dialtone stops after I press
9, which is not what I was led to believe would happen with the
ignorepat directive.
Problem A: Dialing in. If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context. So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute before dropping the
POTS line. If I pick that SIP phone back up, and dial an outside
number, I can reconnect to the dangling call, which will hear the
tones after the 9... The outside caller will finally get dropped after
about a minute of waiting.
Here's a transcript from the CLI:
(I pick up a phone not on the switch, and call the FXO:
-- Starting simple switch on 'Zap/5-1'
-- Executing Answer(Zap/5-1, ) in new stack
-- Executing GotoIfTime(Zap/5-1,
07:30-16:30|mon-fri|*|*?open|s|1) in new stack
-- Goto (open,s,1)
-- Executing DigitTimeout(Zap/5-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/5-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/5-1, alc01) in new stack
-- Playing 'alc01' (language 'en')
== CDR updated on Zap/5-1
-- Executing Macro(Zap/5-1, stdext|102|SIP/102) in new stack
-- Executing Dial(Zap/5-1, SIP/102|20) in new stack
-- Called 102
(The SIP phone begins ringing)
-- SIP/102-08184fa8 is ringing
-- SIP/102-08184fa8 answered Zap/5-1
(Hang up SIP)
== Spawn extension (macro-stdext, s, 1) exited non-zero on 'Zap/5-1'
in macro 'stdext'
== Spawn extension (macro-stdext, s, 1) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
(Pick up SIP, dial 96653674)
-- Executing Dial(SIP/102-08184808, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-08184808
(I heard the tones on the outside phone, which is still off-hook)
(hung up SIP)
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-08184808'
(one more time, pick up SIP)
-- Executing Dial(SIP/102-08184808, Zap/5/6653674) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Congestion(SIP/102-08184808, ) in new stack
(Got fast-busy, so hung up SIP)
== Spawn extension (internal, 96653674, 2) exited non-zero on
'SIP/102-08184808'
(about now, the outside line went back to dialtone.)
Sometimes, I can repeat that pick-back-up trick three or four times.
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I know this has been long, and wordy...hope someone can help. We're
newbs around here, and trying to get things working. My boss is *very*
impressed with the menus and such I've got set up and working, and we've
used soft phones via VPN and it works great...now we just need our
outside lines working!
Thanks a million!
J. David Bavousett
System Administrator
Abilene Library Consortium
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Re: [asterisk-users] Trixbox problems
On Tuesday 15 May 2007 19:11, Dave Cotton wrote: On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote: Hello, I’m writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) - Sometimes when the user dials out, he hears the ringing tone but the line is already answered and the called party hears his voice while he’s still hearing the ringing tone. How can I investigate those 2 problems in order to find what’s happening ? Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox problems
It has nothing to do with the GUI. Trixbox compiles Zaptel for you and provides them as RPMs for installation. Removing the RPMs and all the configs they leave lying around and compiling from source can be a complicated process, and the Trixbox forums/mailing lists will be better able to help the OP in this case. AR On 5/15/07, Diego Iastrubni [EMAIL PROTECTED] wrote: On Tuesday 15 May 2007 19:11, Dave Cotton wrote: On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote: Hello, I'm writing because we have problems with an asterisk installation (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o telephony traffic (more or less 1 call/2 min.); we are using a TDM400 board, with 3 PSTN lines configured and we have two big issues: - Calls are dropped during conversation (I have a busycount=8 from the initial value that was 4) - Sometimes when the user dials out, he hears the ringing tone but the line is already answered and the called party hears his voice while he's still hearing the ringing tone. How can I investigate those 2 problems in order to find what's happening ? Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [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] Trixbox problems
On Tue, 2007-05-15 at 19:57 +0300, Diego Iastrubni wrote: On Tuesday 15 May 2007 19:11, Dave Cotton wrote: Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today Perhaps so am I. Perhaps the fancy-shmancy GUI is hiding the configs. -- Dave Cotton [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] Mr. Spencer Written
On 5/15/07, Seb Auriol [EMAIL PROTECTED] wrote:
Kristian, I think Cleviton's point was that [lookupdundi] does not appear to
be a macro, and so how can it have arguments if it just a normal context /
extension?
Kind regards,
Sebastian
Sebastian,
Reading the post again I could see it either way. I just couldn't
quite understand the English so I guessed... Not that I am
criticizing Cleviton - his English is much better than my
Portuguese...
nao falo Portuguese - see I don't even know where to put the accents! ;)
Back on topic - using ${ARG1} there doesn't seem right but I haven't
read the article (I'm not going to bother signing up) to make sure.
--
Kristian Kielhofner
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Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at which time I've more than once seen the message from the subject.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
Is there a better/more appropriate place/list to ask this kind of
question?
/Per Jessen, Zürich
--
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Starting at SFr1/month/user - http://www.spamchek.ch/
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Re: [asterisk-users] Outside lines are just not happening...
On 5/15/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Two problems, possibly related:
Here's the configuration...my Asterisk box has a TDM844B in it; port 1-4
are FXS, 5-8 FXOs.
Here are the config files:
/etc/zaptel.conf:
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
loadzone= us
defaultzone = us
/etc/asterisk/zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
canpark=yes
rxgain=0.0
txgain=0.0
context=internal
signalling=fxo_ks
channel = 1-4
I recommend that you put in a group, like group=2
context=external
signalling=fxs_ks
channel = 5-8
A snippet from /etc/asterisk/extensions.conf:
[internal]
ignorepat = 9
if you put in the group, you can dial out via:
exten = _9NXX,1,Dial(Zap/g2/${EXTEN:1}) to start with the lowest
available channel, or
Dial(ZAP/G2/${EXTEN:1}) to start with the
highest available channel.
This will let you make more than one outgoing call at a time.
exten = _9NXX,2,Congestion()
exten = _9NXX,102,Congestion()
The SIP phone is also in the internal context, and other things below
that in the context work just fine on the internal network.
I don't know if it's relevant or not, but dialtone stops after I press
9, which is not what I was led to believe would happen with the
ignorepat directive.
Dial tone is generated by the SIP phone. You'll need to configure it
directly on whatever SIP device you're using. Now, if your analog phones
(like on ports 1-4) stop dial tone, you might need to be concerned.
Problem A: Dialing in. If I call from my cell, the FXO picks right up,
and sends me to the voice menu that I have at the top of the [external]
context. So far so good, but if the SIP that I get in touch with hangs
up, the FXO stays off-hook for more than a minute before dropping the
POTS line. If I pick that SIP phone back up, and dial an outside
number, I can reconnect to the dangling call, which will hear the
tones after the 9... The outside caller will finally get dropped after
about a minute of waiting.
This is normal when dealing with POTS lines. You can try to get disconnect
supervision, try to trick zaptel into guessing what the state of the line
is, but in my experience, it just comes with the territory. Disconnect
supervision is, by far, the best solution, but most telcos stick their
fingers in their ears when it's requested...
That's one of the main reasons we use PRI where it makes sense, and have
people hang up the phones where it doesn't.
snip
Problem B: Dialing out. From the SIP phone, if I dial out, here's the
transcript:
-- Executing Dial(SIP/102-081854e0, Zap/5/6653674) in new stack
-- Called 5/6653674
-- Zap/5-1 answered SIP/102-081854e0
-- Hungup 'Zap/5-1'
== Spawn extension (internal, 96653674, 1) exited non-zero on
'SIP/102-081854e0'
Sometimes (about half the time) the phone I'm calling (in this case, a
cell) will give part of one ring, then report a missed call. The SIP
phone hangs up after about 5 seconds. But not always. The rest of the
time, the SIP phone just eventually (15 or 20 secs) hangs up on its'
own, and the cell never reports a missed call.
I'm not sure on this one. It could be a bad line, the line may not be fully
reset from the previous call, or something completely different.
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RE: [asterisk-users] Outside lines are just not happening...
David: Thanks for your tips...all but one issue, then, solved as best it can be solved. Just the outbound dialing issue, now... --David B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox problems
On Tue, 2007-05-15 at 19:16 +0200, Dave Cotton wrote: Perhaps the fancy-shmancy GUI is hiding the configs. Al Bochter has just told me off list that Trixbox is Asterisk But according to their site trixbox is a complete application platform. When you install trixbox you have a powerful application platform at your fingertips. Products included with trixbox include: * trixbox dashboard * Asterisk(tm) Open Source PBX * FreePBX web management tool * SugarCRM * Munin (via package manager) * HUDLite server/admin (via package manager) * IVRGraph (via package manager) * phpMyAdmin? (via package manager) * Webmin (via package manager) So I still wonder if the GUI hides the configs. As I've mentioned on Talkshoe my GUI is called vi. -- Dave Cotton [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
R: [asterisk-users] Trixbox problems
Just to be clear: 1) I do not use to configure the * config files with trixbox GUI but I manually edit the file 2) from my point of view, the main advantage of trixbox is to have an * installation uprunning in half an hour, then it's up to you use the GUI or manually edit files 3) I did not ask 'what I have to change in my configuration' but the question is different and it's 'how can I troubleshoot the problem'. Troubleshooting it's distribution/GUI independent task so if you don't want to help other people just relax and watch a film on the tv, don't waste your time writing unuseful mails. Regards marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Dave Cotton Inviato: martedì 15 maggio 2007 19.17 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Trixbox problems On Tue, 2007-05-15 at 19:57 +0300, Diego Iastrubni wrote: On Tuesday 15 May 2007 19:11, Dave Cotton wrote: Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today Perhaps so am I. Perhaps the fancy-shmancy GUI is hiding the configs. -- Dave Cotton [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: R: [asterisk-users] Trixbox problems
If you use edit the config files on a trixbox system like you would on an * box, any time you reboot or hit the red update bar, it will reset the files to what the gui has. The only files you can edit on a trixbox system are the _custom.conf files. This may be the issue with the time out Martin D. Marco Vescovi wrote: Just to be clear: 1) I do not use to configure the * config files with trixbox GUI but I manually edit the file 2) from my point of view, the main advantage of trixbox is to have an * installation uprunning in half an hour, then it's up to you use the GUI or manually edit files 3) I did not ask 'what I have to change in my configuration' but the question is different and it's 'how can I troubleshoot the problem'. Troubleshooting it's distribution/GUI independent task so if you don't want to help other people just relax and watch a film on the tv, don't waste your time writing unuseful mails. Regards marco -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Dave Cotton Inviato: martedì 15 maggio 2007 19.17 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Trixbox problems On Tue, 2007-05-15 at 19:57 +0300, Diego Iastrubni wrote: On Tuesday 15 May 2007 19:11, Dave Cotton wrote: Contact the Trixbox mailing lists? Why is that? You think some fancy-shmancy GUI will fix this? The problem is obviously in the zaptel area. But hey... this is asterisk-users... /me is in a fighting mode today Perhaps so am I. Perhaps the fancy-shmancy GUI is hiding the configs. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [asterisk-users] Trixbox problems
On 5/15/07, Marco Vescovi [EMAIL PROTECTED] wrote: Just to be clear: 1) I do not use to configure the * config files with trixbox GUI but I manually edit the file 2) from my point of view, the main advantage of trixbox is to have an * installation uprunning in half an hour, then it's up to you use the GUI or manually edit files 3) I did not ask 'what I have to change in my configuration' but the question is different and it's 'how can I troubleshoot the problem'. Troubleshooting it's distribution/GUI independent task so if you don't want to help other people just relax and watch a film on the tv, don't waste your time writing unuseful mails. Regards marco If you've edited the files directly, then you undoubtedly know why you're getting the responses you are getting. There's a few different files that all come together to form each configuration file. And sometimes it's not easy to see what will override others. Another challenge when dealing with trixbox installs is dealing with the permissions. Trixbox, rightly in my opinion, changes who owns files from the default root:root. You just need to be careful when you start monkeying around with the installation files. Dropped calls are not usually easy to narrow down. You need to make sure the line is good. You need to make sure that some of the more esoteric options in zapata.conf are turned off, as I've seen them cause problems. And, frankly, we've had problems out of the 400-series cards, so we only use them in low-traffic areas. But checking IRQ misses, your hard drive DMA settings, and all of the standard troubleshooting techniques may help. I've had the ringing problem before, but for me it was an indications problem. But another time, I always had to use an Answer() before dialing out the Zap interface. Since you're headed to a POTS line anyway, it will be Answer()'d as soon as the call is dialled anyway, and putting the Answer() before the Dial() gave some of my more clueless SIP UAs a hint as to what's going on. Otherwise, they'd disconnect after 1 minute of ringing and leave the line out off-hook. So, the answer to your question is this: do the normal troubleshooting steps. But, since it very well could be configuration-related, you may want to try the Trixbox list, as this may have come up on other installations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail volume
Try the 'g' option to VoiceMail(). CP Stephen Bosch wrote: Hi: I have a user saying that the volume of voice mails is too low. Is there a way to tweak the recording level for voice mail? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox problems
Dave Please note what the core is.. * Asterisk(tm) Open Source PBX The GUI only writes some of the conf file for you. So if there is a fix for the list member that works on Asterisk please help them out. I have worked on other Asterisk based PBX systems and the conf files are just about the same. I am not saying Trixbox is better just easier for the new guy Me I don't like GUI's I prefer the hard way. That way I know what conf files do what to the system and that makes it easer to fix latter. * HUDLite server/admin (via package manager) Just slows the systems down and I see no good use for HUDLite Yes Trixbox does have alot of USELESS Packages added on to it. But keep in mind it is still Asterisk based at the core. The bottom line is. - Trixbox is still [asterisk-users] Best regards, Al Bochter Bochter Services Did you check your US Greenbacks for GOLD Today? http://www.bochterservices.com/?t=USbill_email Dave Cotton wrote: On Tue, 2007-05-15 at 19:16 +0200, Dave Cotton wrote: Perhaps the fancy-shmancy GUI is hiding the configs. Al Bochter has just told me off list that Trixbox is Asterisk But according to their site trixbox is a complete application platform. When you install trixbox you have a powerful application platform at your fingertips. Products included with trixbox include: * trixbox dashboard * Asterisk(tm) Open Source PBX * FreePBX web management tool * SugarCRM * Munin (via package manager) * HUDLite server/admin (via package manager) * IVRGraph (via package manager) * phpMyAdmin? (via package manager) * Webmin (via package manager) So I still wonder if the GUI hides the configs. As I've mentioned on Talkshoe my GUI is called vi. ___ --Bandwidth 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] Zaptel 1.4.2.1 and TE212P
Hi, I have purchased a TE212P (Dual Span) Digium card and have compiled Zaptel 1.4.2.1 and LibPri 1.4.0 and Asterisk 1.4.4 ztcfg -vv shows this: Zaptel Version: 1.4.2.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: Clear channel (Default) (Slaves: 32) Channel 33: Clear channel (Default) (Slaves: 33) Channel 34: Clear channel (Default) (Slaves: 34) Channel 35: Clear channel (Default) (Slaves: 35) Channel 36: Clear channel (Default) (Slaves: 36) Channel 37: Clear channel (Default) (Slaves: 37) Channel 38: Clear channel (Default) (Slaves: 38) Channel 39: Clear channel (Default) (Slaves: 39) Channel 48: D-channel (Default) (Slaves: 48) 40 channels configured. /etc/zaptel.conf consists of: # Config for a UK Euro-ISDN line span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,crc4 bchan=32-39 dchan=48 loadzone=uk defaultzone=uk and /etc/asterisk/zapata.conf consists of: [channels] callprogress=no usecallerid=yes language=en echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn usecallingpres=yes group=1 context=incoming busydetect=no resetinterval=never channels = 1-15,17-31,32-39 The red light on Span 1 Span is flashing and both are showing RED alarms ?? What am I doing wrong ? I have successfully built other Asterisk boxes with different cards in without problems - this is an upgrade for a client. It is being connected to a UK ISDN30 circuit at the weekend 1 with 30 lines the other with 8 lines (38 Lines in total) Any help would be really, really appreciated. Regards Matt Brown [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] function_db_read: DB requires an argument, DB(family/key)
On Tue, 15 May 2007, Per Jessen wrote:
Per Jessen wrote:
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk -
at which time I've more than once seen the message from the subject.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
Is there a better/more appropriate place/list to ask this kind of
question?
Probably not, but ...
You're getting the error message because ${CALLERID(num)} is empty. ie.
there is no caller-Id set, so I'd work on working out why there's no
callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I do omething similar, but I test for no callerId before trying to do a
database lookup.
Gordon
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Re: [asterisk-users] Asterisk High-Capacity Stability
On May 14, 2007, at 1:55 PM, Daryl Jurbala wrote: On May 14, 2007, at 1:29 PM, Zoa wrote: Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). That's great if you're good enough/have the time to make that happen. But when I have issues and call/pay Digium and don't get timely or meaningful answers, it's doesn't make for a good business decision to continue using it for that purpose when I can toss in a Nextone or Sansay and have it just work. All the time. No babysitting. Full professional and timely problem resolution from the vendor, etc, etc, etc. Don't even get me started on Digium not being able to get TC400Bs to properly negotiate g.723.1 5.3k when a client requests 6.3k first (thank god for Cantata). Actually, I suspect that maybe a flaw in Asterisk, since, IIRC, it does not know the difference between 5.3 and 6.3 kbps g.723. IIRC, in Asterisk, the G.723 that it knows about is just one or the other, not both. Though I would like to apologize for your delay, the people responsible for that product are working hard on trying to get the G.723 possibility resolved. Matthew Fredrickson ___ --Bandwidth 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] Astsee v0.1 released - an Asterisk channel monitor for linux/X windows
Hiya everyone. I have been working on a fun little app to watch what's going on in your asterisk box via its manager interface. There's a screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it requires allegro, but I was more keen about getting the ideas down than worrying about the framework. Comments/questions welcome, but probably off-list is best unless they are usage questions that would benefit the community. Source isn't available currently, just a binary. I expect to release the source too, soon, but don't know for sure yet. There are a plethora of inadequacies rampant in this thing, YMMV, details about what works and what doesn't are on the page linked to above. This is more of a proof-of-concept release than even v0.1. Thanks for reading. I hope this provides a viable alternative to the great but seeming-to-be-not-updated gastman software. Moj ___ --Bandwidth and Colocation provided 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 1.4.2.1 and TE212P
Are you sure that you set the T1/E1 jumpers on the board correctly for E1 mode? Matthew Fredrickson On May 15, 2007, at 1:23 PM, Matt Brown wrote: Hi, I have purchased a TE212P (Dual Span) Digium card and have compiled Zaptel 1.4.2.1 and LibPri 1.4.0 and Asterisk 1.4.4 ztcfg -vv shows this: Zaptel Version: 1.4.2.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: Clear channel (Default) (Slaves: 32) Channel 33: Clear channel (Default) (Slaves: 33) Channel 34: Clear channel (Default) (Slaves: 34) Channel 35: Clear channel (Default) (Slaves: 35) Channel 36: Clear channel (Default) (Slaves: 36) Channel 37: Clear channel (Default) (Slaves: 37) Channel 38: Clear channel (Default) (Slaves: 38) Channel 39: Clear channel (Default) (Slaves: 39) Channel 48: D-channel (Default) (Slaves: 48) 40 channels configured. /etc/zaptel.conf consists of: # Config for a UK Euro-ISDN line span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,crc4 bchan=32-39 dchan=48 loadzone=uk defaultzone=uk and /etc/asterisk/zapata.conf consists of: [channels] callprogress=no usecallerid=yes language=en echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn usecallingpres=yes group=1 context=incoming busydetect=no resetinterval=never channels = 1-15,17-31,32-39 The red light on Span 1 Span is flashing and both are showing RED alarms ?? What am I doing wrong ? I have successfully built other Asterisk boxes with different cards in without problems - this is an upgrade for a client. It is being connected to a UK ISDN30 circuit at the weekend 1 with 30 lines the other with 8 lines (38 Lines in total) Any help would be really, really appreciated. Regards Matt Brown [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] cpu usuage
On May 15, 2007, at 1:50 PM, Khaled Chehab wrote: Do any one knows the formula to calculate memory and cpu usuage for channel on g729 codec,to know the hardware required for 100 concurrent call. I believe you can for sure do nearly 100 calls (on pretty much any decent system) with the new transcoder card. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astsee v0.1 released - an Asterisk channel monitor for linux/X windows
I must clarify, when I experimented with gastman, it was on the Windows platform only. Thus, I can understand why it didn't seem to be updated very often. Now that I have entirely switched over to *nix, I have NOT experimented with gastman. I'm sure it has had the opportunity to far surpass my expectations I've learned by working with the Windows version :) No offense meant to the maintainers of the gastman project! Moj Mojo with Horan Company, LLC wrote: Hiya everyone. I have been working on a fun little app to watch what's going on in your asterisk box via its manager interface. There's a screenshot up and some info at http://sitkavoip.com/astsee/ -- Sorry it requires allegro, but I was more keen about getting the ideas down than worrying about the framework. Comments/questions welcome, but probably off-list is best unless they are usage questions that would benefit the community. Source isn't available currently, just a binary. I expect to release the source too, soon, but don't know for sure yet. There are a plethora of inadequacies rampant in this thing, YMMV, details about what works and what doesn't are on the page linked to above. This is more of a proof-of-concept release than even v0.1. Thanks for reading. I hope this provides a viable alternative to the great but seeming-to-be-not-updated gastman software. Moj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feasibility Request
I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). I'm located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Gordon Henderson wrote:
You're getting the error message because ${CALLERID(num)} is empty.
ie. there is no caller-Id set, so I'd work on working out why there's
no callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I do omething similar, but I test for no callerId before trying to do
a database lookup.
Later on I have no problems with e.g. a suppressed callerid - but I'll
try what you suggest. Thanks Gordon.
OK, tried it -
with your Noop(), I don't get a warning when there is no CLIP:
-- Executing [EMAIL PROTECTED]:1] NoOp(mISDN/3-u0, CallerId is
) in new stack
-- Executing [EMAIL PROTECTED]:2] Ringing(mISDN/3-u0, ) in new stack
What surprises is that my Set() call isn't listed in the console log?
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and then
issue the no argument warning on the Set() call.
And miraculously, I can make the whole thing work by issuing a
module reload app_queue. After that, the DB() function no longer
complains, with or without CLIP.
Sounds like a bug to me.
/Per Jessen, Zürich
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Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
Vincent Delporte wrote: Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). = When a call comes in, I'd like an AGI application to send an e-mail and send CID name/number to a script on a web server. Is this the correct way to do it in Perl, with the modules available in AsteriskWin32? Could I rewrite this in Delphi instead? Vincent, Freepascal seems to work very nicely. However, I'm not sure how delphi behaves with stdin/stdout since I've not written many console apps in delphi, mostly GUI rich software. The best bet would be as another poster suggested and to write a FastAGI server. If you don't feel like writing a FastAGI server, you can also take a look at AsterPas which is a pascal based FastAGI server for both Windows and Linux (written in freepascal): http://www.datatrakpos.com/pos/datatalk/asterpas.aspx It's still in beta, but works very well. You can check out some scripts that I've written so far: http://www.leebo.dreamhosters.com/apscripts/ -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feasibility Request
On Tuesday 15 May 2007 3:31 pm, Jeremy Mann wrote: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. I am doing this right now with our MICS. Asterisk is the telco, and routes the calls over our PRI or VOIP provider. I also do a little bit of external extensions. While it works, it's hokey. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. Yes, I was doing this before the PRI. Make sure you're using the right channel bank for FXO, or you won't get CPD. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). Piece of cake, it's just LCR and failover. With the right dialplan nobody knows whether the call went over VOIP or local PSTN. I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. If you want 5 nines out of Asterisk, you're looking at a failover system with a database backend, and T1 failover to the Asterisk boxes. Now you'll also need redundant power and really look at the entire system to make sure there aren't any single points of failure that aren't five nines themselves (i.e. you won't need two PRIs, as they're already considered five nines). Honestly though... take a look at the Citel gateways. Plug all of your Norstar phones into that and connect it to Asterisk. There's your PBX. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). Send me some more information offlist and I'll see what I can do for bidding. Honestly though you'll want to be hands-on on this, as it'll be your butt on the line when (not if) they fall over. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
Vincent Delporte wrote: Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). = When a call comes in, I'd like an AGI application to send an e-mail and send CID name/number to a script on a web server. Is this the correct way to do it in Perl, with the modules available in AsteriskWin32? Could I rewrite this in Delphi instead? Vincent, Freepascal seems to work very nicely. However, I'm not sure how delphi behaves with stdin/stdout since I've not written many console apps in delphi, mostly GUI rich software. The best bet would be as another poster suggested and to write a FastAGI server. If you don't feel like writing a FastAGI server, you can also take a look at AsterPas which is a pascal based FastAGI server for both Windows and Linux (written in freepascal): http://www.datatrakpos.com/pos/datatalk/asterpas.aspx It's still in beta, but works very well. You can check out some scripts that I've written so far: http://www.leebo.dreamhosters.com/apscripts/ We're planning on adding a mail client class to the scripting engine soon. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feasibility Request
On 5/15/07, Jeremy Mann [EMAIL PROTECTED] wrote: I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. Yes, I've already done it. Just make sure you use a T1 cross-over and get the signalling correct (use pri_net instead of pri_cpe) 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. I've done that too, using the same PRI as part 1. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. This isn't really a big deal. Just have a fall-through when PSTN lines are full/down. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). You can get local presence by having a provider who can sell you a DID from your local areas and trunk them to a PRI/T1 in another area, or deliver them over SIP. The challenge with having only one analog line in a city means you can't receive 2 calls at the same time... definitely sub-optimal! I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. We're healthcare too, but in Ophthalmology. So 5-9's aren't really required here, although we've had it. I haven't really had any problems with Asterisk reliability. In the setup you propose, you're probably going to see more challenges in keeping your Internet connections up with good latency than a well-built Asterisk system. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). Buy decent servers, with redundant power supplies, raid-5 arrays with a software mirror across different array controllers, keep a warm-standby at each location, install separate diesel generators in each location, move your offices into underground bunkers in secret, nondescript locations, hire armed trolls to guard the server and pummel anyone who attempts to approach, etc. The point is, you can have as much reliability as you're willing to buy. I'm located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. Spent a couple of years in Addison, and I grew up in Houston. But I can't really offer too much on-location help, as I've moved to FL. Ah well, can't win 'em all, right? But if you get the trolls, I may be willing to make the trip ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 08:21, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: If you think your ISP is reliable enough then go for it! I've had less ADSL issues last year than ISDN issues! ;-) (And that while ADSL is running over that very ISDN line!) There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. It can work out a lot cheaper than going down the traditional ISDN2/ISDN30 route for a lot of people as a small business expands. I can see that would work out that way, yes! Undfortunately I'll have to pay reconnection fee before I can cancel! :-o I guess that's a country thing - good luck :) I found out that I can even transfer my current main number to my ISP's SIP service for EUR 5 a month... Aside from that they can give me 2 free incoming numbers in the 087 range, and I already have an incoming VoipBuster number in my own areacode... That would give me 4 incoming numbers... The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 10:31, Chris Bagnall wrote: There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. Indeed, many of our clients are doing just that. I would, however, strongly recommend against ditching PSTN entirely (in the UK, it's virtually impossible anyway since ADSL requires a PSTN line over which to run) - those PSTN lines are still useful for things like emergency service calls, directory enquiries, etc. etc. In NL you actually can ditch the telephony and keep the ADSL... My ISP even gives emergency access if you transfer your main number to their SIP service. And there still is my cell-phone too! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] socket_process: Received mini frame before first full voice frame
Never received a response for this from anyone. This is being seen more frequently now. Please Suggest. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Cc: asterisk-dev [EMAIL PROTECTED] Sent: Friday, May 11, 2007 2:26:30 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] socket_process: Received mini frame before first full voice frame Anyone any idea why do we keep on getting chan_iax2.c:7535 socket_process: Received mini frame before first full voice frame 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! Havn't doen that in years - an online fax service sends me my faxes by email and I sent out faxes through them, not that I ever do that. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] function_db_read: DB requires an argument, DB(family/key)
Per Jessen wrote:
Gordon Henderson wrote:
You're getting the error message because ${CALLERID(num)} is empty.
ie. there is no caller-Id set, so I'd work on working out why there's
no callerId set for the very first call...
Eg. start with:
exten = _X.,1,Noop(CallerId is ${CallerId(all)})
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I do omething similar, but I test for no callerId before trying to do
a database lookup.
Later on I have no problems with e.g. a suppressed callerid - but I'll
try what you suggest. Thanks Gordon.
OK, tried it -
with your Noop(), I don't get a warning when there is no CLIP:
-- Executing [EMAIL PROTECTED]:1] NoOp(mISDN/3-u0, CallerId is
) in new stack
-- Executing [EMAIL PROTECTED]:2] Ringing(mISDN/3-u0, ) in new stack
What surprises is that my Set() call isn't listed in the console log?
OK, so I tried this:
exten = _X.,1,Noop(CallerId is ${CALLERID(all)})
exten = _X.,n,Noop(blurp)
exten = _X.,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
This now appears to execute the first Noop(), skip the second, and then
issue the no argument warning on the Set() call.
Try an Answer() first?
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Lee
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Re: [asterisk-users] How to write data to astdb?
Vincent Delporte wrote:
Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the first line,
and just waits:
rem c:\cygroot\mystuffimport.bat
rem
rem c:\cygroot\mystuffC:\cygroot\bin\asterisk.exe -rx 'database put
cidname 123 My cellphone'
rem
rem Asterisk module loaded successfully
rem Asterisk entry point foundW2003*CLI Updated database successfully
rem Verbosity is at least 1
rem STUCK HERE!
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 My cellphone'
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 456 This is a test'
I don't know why the batch script stops after the first line.
So, I installed ActivePerl and the asterisk-perl package from CPAN, and
tried this, but it doesn't work either:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-database_put('cidname', '', 'my number');
Is there a way to access astdb directly, instead of through an AGI script?
Thank you.
There are built in Asterisk commands for this:
http://www.voip-info.org/wiki/view/Asterisk+database
--
Warm Regards,
Lee
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[asterisk-users] RE: Mr. Spencer Written
Mr. Spencer written the article Using DUNDi with a Cluster of Asterisk
Servers http://www.voip-magazine.com/content/view/3644/0/1/0/ in the
VoIP Magazine and the piece follow:
[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv
exten = i,1,Goto(lookupmysql,${INVALID_EXTEN},1)
I didn't get understand the usage ARG1 argument in the context. Do use
ARG1 variable only into macro? Is not this usage apparent contradiction
with Asterisk documents? Do anyone get explain this?
Exten = _X,1,Goto should have actually been
Exten = _X.,1,Goto
You can disregard this exten, it was meant to be a catch-all in this
context, but the switch statement is really what is invoked when you send a
call to this context, not the exten = _X.
JR
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Re: [asterisk-users] socket_process: Received mini frame before first full voice frame
Sanjay Rajdev wrote: Never received a response for this from anyone. This is being seen more frequently now. Please Suggest. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Cc: asterisk-dev [EMAIL PROTECTED] Sent: Friday, May 11, 2007 2:26:30 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] socket_process: Received mini frame before first full voice frame Anyone any idea why do we keep on getting chan_iax2.c:7535 socket_process: Received mini frame before first full voice frame It means that the SECOND audio packet for the call arrived before the FIRST audio packet. I suspect that mini-frames are much smaller than the first frame/packet. Some routers may prioritize small packets over larger packets. This should be covered multiple times in the mailing list archives. ___ --Bandwidth and Colocation provided 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 write data to astdb?
I use asterisk -rx database put value if you are trying to batch it
from windows you can use plink
On 5/15/07, Vincent Delporte [EMAIL PROTECTED] wrote:
Hello,
I'm trying to fill CID data into the astdb using AsteriskWin32's
asterisk.exe, to no avail: The batch file stops after the first line, and
just waits:
rem c:\cygroot\mystuffimport.bat
rem
rem c:\cygroot\mystuffC:\cygroot\bin\asterisk.exe -rx 'database put
cidname 123 My cellphone'
rem
rem Asterisk module loaded successfully
rem Asterisk entry point foundW2003*CLI Updated database successfully
rem Verbosity is at least 1
rem STUCK HERE!
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 My cellphone'
C:\cygroot\bin\asterisk.exe -rx 'database put cidname 456 This is a test'
I don't know why the batch script stops after the first line.
So, I installed ActivePerl and the asterisk-perl package from CPAN, and
tried this, but it doesn't work either:
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$AGI-database_put('cidname', '', 'my number');
Is there a way to access astdb directly, instead of through an AGI script?
Thank you.
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[asterisk-users] RE: Web based call control
There's a better way. Take a look at how to do a Find me at http://www.voip-info.org/wiki/view/Asterisk+tips+findme You can have the call only completed when they press a key on the receiving phone. No voicemail will trigger that. Chip Schweiss -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Monday, May 14, 2007 10:45 PM To: asterisk-users@lists.digium.com Subject: Web based call control Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will take to go to their home phone voicemail, but I don't want to deliver the call there, I want it to go to the next priority in asterisk. So I was thinking that it would be nice to build a web interface that they could have a button to answer with. This would send a manager command to the server telling it to answer the channel, any thoughts on how to do this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feasibility Request
Jeremy, Both 1 and 2 are feasible and have been done by many people including the company I currently work for and the company I previously worked for. For the analog lines, I would recommend a channel bank with analog ports. If you want to redirect inbound calls on an analog line as well as send calls via analog to a PBX to support 12 lines you will need 24 ports. If you are only using 12 ports a channel bank may not prove to be cost effective. If you use a channel bank then the hardware for system 1 and system 2 could be the same exact system. -Jonathan Jeremy Mann wrote: I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1. Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I’d pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I’d be using anywhere from 4-12 lines depending on location size. I’d like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. a. I’d also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I’d attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I’d then beef up my T1 locations to handle more VoIP based calls. Currently we’re using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). I’d also like any insight or suggestions on uptime. We’re a healthcare organization so 5-9’s is what we’ll require. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don’t need VoIP capable phones yet, but if the system works well enough we’d probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9’s is critical). I’m located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.4 reproducibly dumps core on Solaris 10
I have built Asterisk 1.4.4 on my Solaris 10 x86 box: LDFLAGS='-R/usr/sfw/lib -R/opt/csw/lib -L/opt/csw/lib -L/usr/sfw/lib' CPPFLAGS=-I/opt/csw/include ./configure -with-curl=/opt/csw --without-oss --without-vpb --prefix=/opt/asterisk-1.4 The build and install go fine but the asterisk executable reproducibly dumps core with a segmentation violation. If I start it as: asterisk -gc and then type help I get a core dump. Using gdb shows the following trace. I get the same trace everytime. # gdb ./asterisk ./core GNU gdb 6.2.1 Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i386-pc-solaris2.10... Core was generated by `./asterisk -gc'. Program terminated with signal 11, Segmentation fault. Reading symbols from /usr/lib/libcurses.so.1...done. Loaded symbols for /usr/lib/libcurses.so.1 Reading symbols from /usr/lib/libpthread.so.1... warning: Lowest section in /usr/lib/libpthread.so.1 is .dynamic at 0074 done. Loaded symbols for /usr/lib/libpthread.so.1 Reading symbols from /usr/lib/libdl.so.1...done. Loaded symbols for /usr/lib/libdl.so.1 Reading symbols from /usr/lib/libnsl.so.1...done. Loaded symbols for /usr/lib/libnsl.so.1 Reading symbols from /usr/lib/libsocket.so.1...done. Loaded symbols for /usr/lib/libsocket.so.1 Reading symbols from /usr/lib/libresolv.so.2...done. Loaded symbols for /usr/lib/libresolv.so.2 Reading symbols from /usr/sfw/lib/libstdc++.so.6...done. Loaded symbols for /usr/sfw/lib/libstdc++.so.6 Reading symbols from /usr/lib/libm.so.2...done. Loaded symbols for /usr/lib/libm.so.2 Reading symbols from /usr/sfw/lib/libgcc_s.so.1...done. Loaded symbols for /usr/sfw/lib/libgcc_s.so.1 Reading symbols from /usr/lib/libc.so.1...done. Loaded symbols for /usr/lib/libc.so.1 Reading symbols from /opt/asterisk/lib/modules/res_musiconhold.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_musiconhold.so Reading symbols from /opt/asterisk/lib/modules/res_adsi.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_adsi.so Reading symbols from /opt/asterisk/lib/modules/res_config_pgsql.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_config_pgsql.so Reading symbols from /usr/lib/libpq.so.4...done. Loaded symbols for /usr/lib/libpq.so.4 Reading symbols from /usr/lib/libz.so...done. Loaded symbols for /usr/lib/libz.so Reading symbols from /usr/sfw/lib/libssl.so.0.9.7...done. Loaded symbols for /usr/sfw/lib/libssl.so.0.9.7 Reading symbols from /usr/sfw/lib/libcrypto.so.0.9.7...done. Loaded symbols for /usr/sfw/lib/libcrypto.so.0.9.7 Reading symbols from /usr/sfw/lib/libssl_extra.so.0.9.7...done. Loaded symbols for /usr/sfw/lib/libssl_extra.so.0.9.7 Reading symbols from /usr/sfw/lib/libcrypto_extra.so.0.9.7...done. Loaded symbols for /usr/sfw/lib/libcrypto_extra.so.0.9.7 Reading symbols from /opt/asterisk/lib/modules/res_features.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_features.so Reading symbols from /opt/asterisk/lib/modules/res_indications.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_indications.so Reading symbols from /opt/asterisk/lib/modules/res_monitor.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_monitor.so Reading symbols from /opt/asterisk/lib/modules/res_smdi.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_smdi.so Reading symbols from /opt/asterisk/lib/modules/res_snmp.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_snmp.so Reading symbols from /usr/sfw/lib/libnetsnmpagent.so.5...done. Loaded symbols for /usr/sfw/lib/libnetsnmpagent.so.5 Reading symbols from /usr/sfw/lib/libnetsnmpmibs.so.5...done. Loaded symbols for /usr/sfw/lib/libnetsnmpmibs.so.5 Reading symbols from /usr/sfw/lib/libnetsnmphelpers.so.5...done. Loaded symbols for /usr/sfw/lib/libnetsnmphelpers.so.5 Reading symbols from /usr/sfw/lib/libnetsnmp.so.5...done. Loaded symbols for /usr/sfw/lib/libnetsnmp.so.5 Reading symbols from /usr/lib/libkvm.so.1...done. Loaded symbols for /usr/lib/libkvm.so.1 Reading symbols from /usr/lib/libpkcs11.so.1...done. Loaded symbols for /usr/lib/libpkcs11.so.1 Reading symbols from /usr/lib/libkstat.so.1...done. Loaded symbols for /usr/lib/libkstat.so.1 Reading symbols from /usr/lib/libelf.so.1...done. Loaded symbols for /usr/lib/libelf.so.1 Reading symbols from /usr/lib/libadm.so.1...done. Loaded symbols for /usr/lib/libadm.so.1 Reading symbols from /usr/lib/libcryptoutil.so.1...done. Loaded symbols for /usr/lib/libcryptoutil.so.1 Reading symbols from /usr/lib/libdoor.so.1...done. Loaded symbols for /usr/lib/libdoor.so.1 Reading symbols from /opt/asterisk/lib/modules/res_speech.so...done. Loaded symbols for /opt/asterisk/lib/modules/res_speech.so Reading symbols from
[asterisk-users] [RTP] PSTN - Gateway - Phone
Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or do they go through the PBX, ie. is it... POST - VoIP gateway - NAT - Net - NAT - IP phone or POST - VoIP gateway - PBX - NAT - Net - NAT - IP phone ? 2. Regardless of the route RTP packets take, do I have to map ports on both NAT routers for RTP packets to be let inside the LAN, or is STUN able to handle this itself? How do I know if the routers are STUN-friendly, or I have to map ports? 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
[asterisk-users] PATH_MAX' undeclared here (not in a function) in asterisk!
hello, James FitzGibbon: thank you for your help. i am very new to arm-linux and embedded linux. i think what you said is right. i am not very sure the steps i taken are correct. i post it here and please give me some help. it might be help other arm-linux users too. i installed all necessary libraries in my linux. if i just install asterisk under my linux. there is no problem. but when i cross-compile it, it has errors. i follow this url:http://wiki.neurostechnology.com/index.php/Asterisk, which tell me how to install asterisk in arm-linux. the steps are: 1. Download the asterisk source 1.4. 2. To the source directory ./configure --build=i686-linux --host=arm-linux --without-pwlib --without-curl --prefix=/opt/OSD/neuros-bsp/toolchain/arm-linux 3. # Look for the line in the Makefile, ifeq ($(ASTDATADIR),) ASTDATADIR:=$(ASTVARLIBDIR) endif Add the below lines, after the above lines(in Makefile), ASTETCDIR=./kasterisk/etc ASTLIBDIR=./kasterisk/lib ASTVARLIBDIR=./kasterisk/var ASTSPOOLDIR=./kasterisk/spool ASTLOGDIR=./kasterisk/log ASTHEADERDIR=./kasterisk/include ASTBINDIR=./kasterisk/bin ASTSBINDIR=./kasterisk/sbin ASTVARRUNDIR=./kasterisk/var/run ASTMANDIR=./kasterisk/man ASTDATADIR:=$(ASTVARLIBDIR) 4. Edit the makeopts file, (For somereason STRIP is not set by the configure, change the line of strip as follows, STRIP=arm-linux-strip 5. make menuselect 6. Select 'Module Embedding' [*] 1. apps [*] 2. cdr [*] 3. channels [*] 4. codecs [*] 5. formats [*] 6. funcs [*] 7. pbx [*] 8. res Select the applications using space bar. 7. In the mainmenu, select Compiler Flags, [*]13. STATIC_BUILDSince we are going to do a static build. 8. make 9. Now go the Makefile of each subdirectories compiled, Prepend the following line with ., $(DESTDIR)$(MODULES_DIR) need to get as .$(DESTDIR)$(MODULES_DIR) 10: make install i think asterisk should link with arm-linux-kernel, but it does not. please give me hints for that problem. thanks! zhu - 抢注雅虎免费邮箱3.5G容量，20M附件！ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is not showing the correct Incomming CallerID
Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the problem is either in misdn.conf or extension.conf. Would be kind enough if some give any Pointer or help. Regards Farooq -- ___ --Bandwidth and Colocation provided 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 is not showing the correct Incomming CallerID
I forgot to give the asterisk logs pbx*CLI -- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack -- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack -- Executing LookupBlacklist(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new stack -- Executing Return(mISDN/2-2, ) in new stack -- Executing Goto(mISDN/2-2, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(mISDN/2-2, user-callerid|) in new stack -- Executing NoOp(mISDN/2-2, user-callerid: 1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 0?report) in new stack -- Executing GotoIf(mISDN/2-2, 0?start) in new stack -- Executing Set(mISDN/2-2, REALCALLERIDNUM=1416222888) in new stack -- Executing NoOp(mISDN/2-2, REALCALLERIDNUM is 1416222888) in new stack -- Executing Set(mISDN/2-2, AMPUSER=) in new stack -- Executing Set(mISDN/2-2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(mISDN/2-2, 1?report) in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp(mISDN/2-2, TTL: ARG1: ) in new stack -- Executing GotoIf(mISDN/2-2, 0?continue) in new stack -- Executing Set(mISDN/2-2, _TTL=64) in new stack -- Executing GotoIf(mISDN/2-2, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(mISDN/2-2, Using CallerID 1416222888) in new stack -- Executing Set(mISDN/2-2, modifiedcallerid=1416222888) in new stack -- Executing Set(mISDN/2-2, CALLERID(number)=1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 1?skipdb) in new stack -- Goto (ext-group,1,4) -- Executing Set(mISDN/2-2, __NODEST=) in new stack -- Executing Set(mISDN/2-2, __BLKVM_OVERRIDE=BLKVM/1/mISDN/2-2) in new stack -- Executing Set(mISDN/2-2, __BLKVM_BASE=1) in new stack -- Executing Set(mISDN/2-2, DB(BLKVM/1/mISDN/2-2)=TRUE) in new stack -- Executing Set(mISDN/2-2, RRNODEST=) in new stack -- Executing Set(mISDN/2-2, __NODEST=1) in new stack -- Executing GotoIf(mISDN/2-2, 1?REPCID) in new stack -- Goto (ext-group,1,14) -- Executing NoOp(mISDN/2-2, CALLERID(name) is ) in new stack -- Executing Set(mISDN/2-2, RecordMethod=Group) in new stack -- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in new stack -- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(mISDN/2-2, recordingcheck|20070516-140757|1179288477.1037) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(mISDN/2-2, No recording needed) in new stack -- Executing Set(mISDN/2-2, RingGroupMethod=hunt) in new stack -- Executing Macro(mISDN/2-2, dial|10||903-909) in new stack -- Executing DeadAGI(mISDN/2-2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '1416222888' dialparties.agi: Methodology of ring is 'hunt' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 903 to extension map -- dialparties.agi: Added extension 909 to extension map -- dialparties.agi: Extension 903 cf is disabled -- dialparties.agi: Extension 909 cf is disabled -- dialparties.agi: Extension 903 do not disturb is disabled -- dialparties.agi: Extension 909 do not disturb is disabled dialparties.agi: extnum: 903 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: extnum: 909 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: NODEST: 1 adding M(auto-blkvm) to dialopts: M(auto-blkvm) -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(mISDN/2-2, Returned from dialparties with hunt groups to dial ) in new stack -- Executing Set(mISDN/2-2, HuntLoop=0) in new stack -- Executing GotoIf(mISDN/2-2, 1?30 ) in new stack -- Goto (macro-dial,s,30) -- Executing Set(mISDN/2-2, HuntMember=HuntMember0) in new stack -- Executing GotoIf(mISDN/2-2, 1?32:35 ) in new stack -- Goto (macro-dial,s,32) -- Executing Set(mISDN/2-2, CT_EXTEN=903) in new stack -- Executing Set(mISDN/2-2, DB(CALLTRACE/903)=1416222888) in new stack -- Executing Goto(mISDN/2-2, s|42) in new stack -- Goto (macro-dial,s,42) -- Executing Dial(mISDN/2-2, SIP/903|10|M(auto-blkvm) ) in new stack -- Called 903 -- SIP/903-08872570 is
RE : [asterisk-users] Asterisk is not showing the correctIncomming CallerID
Hi Farook and the list, You have may be forgotten to input that in the misdn.conf file : nationalprefix=0 internationalprefix=00 dialplan=0 localdialplan=0 cpndialplan=0 Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Farooq Ahmed Envoyé : mercredi 16 mai 2007 06:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk is not showing the correctIncomming CallerID I forgot to give the asterisk logs pbx*CLI -- Executing Set(mISDN/2-2, FROM_DID=3722) in new stack -- Executing Gosub(mISDN/2-2, app-blacklist-check|s|1) in new stack -- Executing LookupBlacklist(mISDN/2-2, ) in new stack -- Executing GotoIf(mISDN/2-2, 0?blacklisted) in new stack -- Executing Return(mISDN/2-2, ) in new stack -- Executing Goto(mISDN/2-2, ext-group|1|1) in new stack -- Goto (ext-group,1,1) -- Executing Macro(mISDN/2-2, user-callerid|) in new stack -- Executing NoOp(mISDN/2-2, user-callerid: 1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 0?report) in new stack -- Executing GotoIf(mISDN/2-2, 0?start) in new stack -- Executing Set(mISDN/2-2, REALCALLERIDNUM=1416222888) in new stack -- Executing NoOp(mISDN/2-2, REALCALLERIDNUM is 1416222888) in new stack -- Executing Set(mISDN/2-2, AMPUSER=) in new stack -- Executing Set(mISDN/2-2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(mISDN/2-2, 1?report) in new stack -- Goto (macro-user-callerid,s,11) -- Executing NoOp(mISDN/2-2, TTL: ARG1: ) in new stack -- Executing GotoIf(mISDN/2-2, 0?continue) in new stack -- Executing Set(mISDN/2-2, _TTL=64) in new stack -- Executing GotoIf(mISDN/2-2, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(mISDN/2-2, Using CallerID 1416222888) in new stack -- Executing Set(mISDN/2-2, modifiedcallerid=1416222888) in new stack -- Executing Set(mISDN/2-2, CALLERID(number)=1416222888) in new stack -- Executing GotoIf(mISDN/2-2, 1?skipdb) in new stack -- Goto (ext-group,1,4) -- Executing Set(mISDN/2-2, __NODEST=) in new stack -- Executing Set(mISDN/2-2, __BLKVM_OVERRIDE=BLKVM/1/mISDN/2-2) in new stack -- Executing Set(mISDN/2-2, __BLKVM_BASE=1) in new stack -- Executing Set(mISDN/2-2, DB(BLKVM/1/mISDN/2-2)=TRUE) in new stack -- Executing Set(mISDN/2-2, RRNODEST=) in new stack -- Executing Set(mISDN/2-2, __NODEST=1) in new stack -- Executing GotoIf(mISDN/2-2, 1?REPCID) in new stack -- Goto (ext-group,1,14) -- Executing NoOp(mISDN/2-2, CALLERID(name) is ) in new stack -- Executing Set(mISDN/2-2, RecordMethod=Group) in new stack -- Executing Macro(mISDN/2-2, record-enable|903-909|Group) in new stack -- Executing GotoIf(mISDN/2-2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(mISDN/2-2, recordingcheck|20070516-140757|1179288477.1037) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(mISDN/2-2, No recording needed) in new stack -- Executing Set(mISDN/2-2, RingGroupMethod=hunt) in new stack -- Executing Macro(mISDN/2-2, dial|10||903-909) in new stack -- Executing DeadAGI(mISDN/2-2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is '1416222888' dialparties.agi: Methodology of ring is 'hunt' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' -- dialparties.agi: Added extension 903 to extension map -- dialparties.agi: Added extension 909 to extension map -- dialparties.agi: Extension 903 cf is disabled -- dialparties.agi: Extension 909 cf is disabled -- dialparties.agi: Extension 903 do not disturb is disabled -- dialparties.agi: Extension 909 do not disturb is disabled dialparties.agi: extnum: 903 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: extnum: 909 dialparties.agi: exthascw: 1 dialparties.agi: exthascfb: 0 dialparties.agi: extcfb: dialparties.agi: exthascfu: 0 dialparties.agi: extcfu: dialparties.agi: NODEST: 1 adding M(auto-blkvm) to dialopts: M(auto-blkvm) -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(mISDN/2-2, Returned from dialparties with hunt groups to dial ) in new stack -- Executing Set(mISDN/2-2, HuntLoop=0) in new stack -- Executing GotoIf(mISDN/2-2, 1?30 ) in new stack -- Goto (macro-dial,s,30)
