Re: [Asterisk-Users] Asterisk and Snom 360
We couldn't set our 1.0.10 Asterisk system to pickup calls with Snom phones. I've read patches http://bugs.digium.com/view.php?id=5014 and http://bugs.digium.com/view.php?id=5853 could provide that with 1.2.X but we never tried ourselves. I would be very happy to know if someone put that in a production system. Regards - Original Message - From: Darrell Long [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 14, 2006 9:28 PM Subject: [Asterisk-Users] Asterisk and Snom 360 Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone 877-777-2932 Direct 914-271-4500 x402 Fax 914-271-4292 ___ --Bandwidth 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] Alcatel 4200 series pbx
Igor Neves [EMAIL PROTECTED] writes: Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Hi Igor, we are doing that. Bristuffed Asterisk with two HFC-cards is running as NT and the Alcatel is CPE. Do you have any specific problem? cu, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Consensus certainly seems to be the Junghanns cards are amongst the best, but not exactly cheap. If you only need to service 2 BRIs, you might want to look at some of the passive options. We have a number of sites here in the UK running 2 HFC-S based cards in a box, all of which seem quite satisfactory (no echo, etc.). Over here you can pick up HFC-S based cards (the ones we use are these: http://www.solwise.co.uk/isdn.htm) for under £20, so they're probably even cheaper on your side of the pond. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
On 2/14/06, Michael Collins [EMAIL PROTECTED] wrote: Nik, I'm not sure that NOP is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have OK under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. ok, with your configuration incoming calls works, but: - i have eco (maybe i have to increase/decrease echotraining value?) - outgoing calls doesn't works ( -- Executing Dial(SIP/102-cc9b, ZAP/g0/mynumber) in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto(SIP/102-cc9b, s-CONGESTION|1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp(SIP/102-cc9b, Dial failed due to CONGESTION) in new stack -- Executing Macro(SIP/102-cc9b, outisbusy) in new stack -- Executing Playback(SIP/102-cc9b, all-circuits-busy-now) in new stack ) it seems that i don't have any channel for outbound - ALARM is set on NOP i've got a TE205P and my zaptel.conf is: span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone= it defaultzone = 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] BRI Newbie - What Hardware, PCI, in the US?
On Wed, 2006-02-15 at 08:47 +, Chris Bagnall wrote: I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Consensus certainly seems to be the Junghanns cards are amongst the best, but not exactly cheap. If you only need to service 2 BRIs, you might want to look at some of the passive options. We have a number of sites here in the UK running 2 HFC-S based cards in a box, all of which seem quite satisfactory (no echo, etc.). Over here you can pick up HFC-S based cards (the ones we use are these: http://www.solwise.co.uk/isdn.htm) for under £20, so they're probably even cheaper on your side of the pond. Same here, we use 2 MRI HFC-S cards in one box. We use bristuff. We had terrible issues with isdn4linux and capi (admittedly that was a year ago, it might be better now). With bristuff it all works very well. conrad ___ --Bandwidth 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] inbound DID trunked
with the following configuration: zapata.conf [channels] language=it context=from-pstn signalling=pri_cpe switchtype=5ess rxwink=300 callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31,32-46,48-62 zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone= it defaultzone = it when i receive a call from a mobile phone, for example from the number 333- to the virtual number 0465-77 i see in the logs call from 333- to 465- if i call from 0465-777888 to 0465-77 i see in the logs call from 465-777888 to 465- why the first 0 is never displayed? and why if the call comes from an ISDN line the DID isn't totally displayed? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel problem on 4 Processor Opteron SMP system
Hi All,
I've just put together a system comprising of the following;
Hardware
2 x AMD Opteron 270 Processors (Dual Core)
Tyan K8WE Mobo
2GB Kingston PC3200 Registered RAM
2 x WD Raptor 1rpm 74Gb
Digium TE210p
Software
Mandriva 2006 Public Release (Kernel 2.6.12-12mdksmp)
Asterisk 1.2.4
Zaptel 1.2.3
Problem
Zaptel compiles and installs to the right place after modifiing /usr/src/linux2.6.12-12/Makefile EXTRAVERSION it then installs to /lib/modules/2.6.12-12mdksmp/misc/.
When I run modprobe zaptel I get FATAL: Error inserting zaptel (/lib/modules/2.6.12-12mdksmp/misc/zaptel.ko): Invalid module format
uname -a reveals;
[EMAIL PROTECTED] zaptel-1.2.3]# uname -a
Linux asteriskpbx 2.6.12-12mdksmp #1 SMP Fri Sep 9 17:20:34 CEST 2005 x86_64 Dual Core AMD Opteron(tm) Processor 270 unknown GNU/Linux
Both the kernel package and the kernel source package are exactly the same version 2.6.12-12mdksmp and 2.6.12-12mdk respectively.
Regards
Chris Teesdale
I.T. / I.P Telephony Development
Philips
Tel : 01325 384394 ex 246
fax : 01325 383876
Email : [EMAIL PROTECTED]
Web : http://www.philips.org.uk
IMPORTANT:
This email and any attachments may be confidential and/or privileged. Everything is intended for use of the addressee only. If you are not the
named addressee you must not disseminate, distribute or copy this email. If you receive this email in error please notify the sender by replying
to this email or by telephoning (+44)(0)870 609 1554 then delete this message from your system. Philips Collection Services Ltd. ("Philips")
routinely monitors the content of email sent and received on its network, to ensure compliance with its policies and procedures. Although
Philips have taken reasonable precautions to ensure no viruses are present in this email or any files attached to it, it cannot accept
any responsibility for any loss or damage arising from the use of this email or its attachments and advises you to carry out appropriate
virus checks. Philips are not responsible for any changes made to the message after it has been sent nor any files attached to it after
it was sent. Emails that contain encrypted material, program files, are obscene, inflammatory, criminal, offensive, in breach of copyright,
contain a virus or threat to computer systems, appear to be a threat to the company or in breach of company policy may be intercepted and/or deleted.
Philips does not accept any liability for any statements made which are clearly the sender's own and not made on behalf of Philips.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] attended call transfer
On Mon, 2006-02-13 at 21:20 -0800, Michael Collins wrote: JCC, So let's consider an operator, takes a call and decides to attended transfer it to Bob because it's slow and she want's to ask something, but the instant she picks that option another call comes in. If hanging up converted it to blind transfer she could get on with her work and answer the next call, as it is she needs to wait till something happens and possibly lose the next call. OK, it's a stretch but it does seem like hanging up the call is just wrong! Absolutely right. I looked at res_features.c and thought maybe I can do a quick fix and invoke app_dial instead of hanging up the channel ;) But that fails because the channel remains locked. I know absolutely nothing about the locks of a channel in asterisk but I'll dig around and in a few years I might be able to fix it ;) Of course if someone has got good recommendations how to properly do it I'm all ears. Conrad ___ --Bandwidth 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] Brief pauses during calls
Hi, I'm experiencing brief pauses during my calls: 0.5-1.0 sec of silence if call continues for more than a few minutes. I'm sure that problem is in the phone (a cheap ATCOM AT-320 with latest SIP firmware) but I'd like to diagnose better. During a little test, it seems that there is no problem with IAX2 firmware (but there are others... I'm not able to transfer and pickup calls...) Can I try different codec/jitterbuffer/othertrick? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering
On 2/7/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: Removing this line will likely fix the problem. Since you don't have a NAT,the qualify= setting doesn't help keep the port(s) open. At the same time,most SIP devices have a NAT Keep Alive option, if that is an issue. HelloIt did fix my problem, thank you for this.Wonder why this use to work with Asterisk earlier than 1.2.x ___ --Bandwidth 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: SIP Register
Subject: RE: SIP Register From: Tomislav Parcina [EMAIL PROTECTED] In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... First impressions telling me you want to check your phone settings. What phone are you using and what are the config settings? Hi Mark, thank you for your reply. I'm using Cisco 7905 with SIP version 1.3.1(050608A). This phone has tone of settings (few pages). What exactly would you need? Why do you think it's phone problem and not Asterisk? Asterisk is the one that contents my provider. * is the one who should decide what information's to send to my VoIP provider... Anyway, I'm inexperienced with this and I'm just trying to understand what is happening and where could be the problem. -- Tomislav Parcina [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] RE: Queue - check agent
Subject: RE: Queue - check agent From: Tomislav Parcina [EMAIL PROTECTED] In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, I might be wrong here, but I thought that in Queues.conf, if you defined a queue with joinempty=no, or joinempty=strict then no calls will be placed in the queue, and asterisk will go onto the next extension in the dial plan. This is fine if it goes to next extension. ; If you wish to remove callers from the queue when new callers cannot join, ; set this setting to one of the same choices for 'joinempty' ; ; leavewhenempty = yes Where the caller goes if last agent exits queue? -- Tomislav Parcina [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] [Announce] Web-MeetMe v2.0.0
Does this work with asterisk 1.2.4?I can't make app_cbmysql work.I get an error when starting asterisk:[app_cbmysql.so]Feb 15 10:26:53 WARNING[7616]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_store_result Feb 15 10:26:53 WARNING[7616]: loader.c:554 load_modules: Loading module app_cbmysql.so failed!Who had a successfull experience compiling/running app_cbmysql.so with asterisk 1.2.4?b.en.q On 1/12/06, Dan Austin [EMAIL PROTECTED] wrote: [New Features]1.Added focus and tab-order to all input fields2.Dynamic generation of date/month/year listboxes a.It is no longer possible to schedule an invalid date. 3.Added 'Extend' and 'End Now' buttons to the monitorpage.4.Invite button on the monitor page.This greatlysimplifies the process of adding callers to a conference. The ./lib/defines file includes definitions for theprefered channel and context***5.Call history report.Support for this featurerequires the php script ./lib/cbEnd.php be running at all times.This also requires a new table in themeetme database if you're upgrading from an earlierrelease.***[Location] http://www.fitawi.com/Asterisk[Files]Web-MeetMe_v2.0.0.tgz (required)app_cbmysql.c (required)cbmysql.conf (required) cb-extensions.conf (suggested)README (suggested)[Installation]See the README[Features]1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blankii. Identify pin # conflicts (another conference withthe same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is setbut the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference3. Separate views for Current, Past and Future conferences4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added7. Users can only monitor, update or delete their conferences8. Verified administrators can monitor, update or delete anyconferences.9. Updated to Asterisk 1.2.0 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that)Thanks and enjoy,Dan***Beta testers and anyone who downloaded v2.0.0 before today The only changes from the beta was a cosmetic change to work withnon-IE browsers and a couple of installation hints.I onlyreceived feedback from one tester, so it appears the package isready to go. ***Developer help/guidence request***The PHP script to monitor conference endtime andup date the CDR is fragile.If Asterisk is shutdown for more than 30 seconds, the script exits.I'd like to make it more resilent.If any PHP experts can make suggests on how to improve thescript it would be appreciated___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Queue - check agent
Maybe to a voicemail message box, which then gets emailed to a special email account. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tomislav Parcina Sent: 15 February 2006 10:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RE: Queue - check agent Subject: RE: Queue - check agent From: Tomislav Parcina [EMAIL PROTECTED] In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, I might be wrong here, but I thought that in Queues.conf, if you defined a queue with joinempty=no, or joinempty=strict then no calls will be placed in the queue, and asterisk will go onto the next extension in the dial plan. This is fine if it goes to next extension. ; If you wish to remove callers from the queue when new callers cannot join, ; set this setting to one of the same choices for 'joinempty' ; ; leavewhenempty = yes Where the caller goes if last agent exits queue? -- Tomislav Parcina [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] RE: SIP Register
Why do you think it's phone problem and not Asterisk? Asterisk is the one that contents my provider. * is the one who should decide what information's to send to my VoIP provider... Anyway, I'm inexperienced with this and I'm just trying to understand what is happening and where could be the problem. One more thing. Now I have tried with softphone. I have the same problem. Asterisk sends user and password of SIP account (SIP phone) that is making a call but not the account information's that I have received from my service provider. Question: How to configure Asterisk so he sends right user information's? -- Tomislav Parcina [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] Alcatel 4200 series pbx
We are using a PRI connection between Asterisk and an Alcatel PBX 4400. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Igor Neves Sent: Monday, February 13, 2006 11:13 AM To: Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Alcatel 4200 series pbx Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
On Tue, Feb 14, 2006 at 03:09:59PM -0700, Chuck Bunn wrote: Hi Giorgio: That seems like a kind of a kludge. I would rather have the program work right, than adding a work around. Dan of Littlejohnsconsulting has told me of one problem in ARI that he is fixing but I do not understand how it will fix the issue yet?? I will let you know as I find out more... A while ago I have submitted a patch to fix that. The part of the patch that clean up voicemail.conf has been accepted. The part of it that changes permissions to group writable has not. http://bugs.digium.com/view.php?id=5929 Please file a feature request to change those two #define-s if you believe that this is useful (I think so). No coding is required. But anyway, even after the change in the code, you'll need to make sure your umask is properly set. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth 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 running on DMZ (no NAT) PROBLEMS- OPTION message is out of State
Hello, Currenly I've [EMAIL PROTECTED] 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: sip reason:Attack Info - Malformed SIP datagram, OPTION message is out of State By the way i've one client that is running all ok, the others all have this problem. I hope some one could help me with this. Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0
The error looks like a problem with the MySQL libraries on your system. I have not tested it against 1.2.4, but do have it running on SVN 7668 and have had it running on 1.2.0 I can try 1.2.4 next week if you are not able to resolve it by them. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben QSent: Wednesday, February 15, 2006 2:46 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0 Does this work with asterisk 1.2.4?I can't make app_cbmysql work.I get an error when starting asterisk:[app_cbmysql.so]Feb 15 10:26:53 WARNING[7616]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_store_result Feb 15 10:26:53 WARNING[7616]: loader.c:554 load_modules: Loading module app_cbmysql.so failed!Who had a successfull experience compiling/running app_cbmysql.so with asterisk 1.2.4?b.en.q On 1/12/06, Dan Austin [EMAIL PROTECTED] wrote: [New Features]1.Added focus and tab-order to all input fields2.Dynamic generation of date/month/year listboxes a.It is no longer possible to schedule an invalid date. 3.Added 'Extend' and 'End Now' buttons to the monitorpage.4.Invite button on the monitor page.This greatlysimplifies the process of adding callers to a conference.The ./lib/defines file includes definitions for theprefered channel and context***5.Call history report.Support for this featurerequires the php script ./lib/cbEnd.php be running at all times.This also requires a new table in themeetme database if you're upgrading from an earlierrelease.***[Location] http://www.fitawi.com/Asterisk[Files]Web-MeetMe_v2.0.0.tgz (required)app_cbmysql.c (required)cbmysql.conf (required) cb-extensions.conf (suggested)README (suggested)[Installation]See the README[Features]1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blankii. Identify pin # conflicts (another conference withthe same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is setbut the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference3. Separate views for Current, Past and Future conferences4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added7. Users can only monitor, update or delete their conferences8. Verified administrators can monitor, update or delete anyconferences.9. Updated to Asterisk 1.2.0 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that)Thanks and enjoy,Dan***Beta testers and anyone who downloaded v2.0.0 before today The only changes from the beta was a cosmetic change to work withnon-IE browsers and a couple of installation hints.I onlyreceived feedback from one tester, so it appears the package isready to go.***Developer help/guidence request***The PHP script to monitor conference endtime andup date the CDR is fragile.If Asterisk is shutdown for more than 30 seconds, the script exits.I'd like to make it more resilent.If any PHP experts can make suggests on how to improve thescript it would be appreciated___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Snom 360
Could we possibly see your settings to get this right? I am trying to get it working at the moment. I can see the phone buttons have subscribed to asterisk, but they just don't light up. We are using 4.1 firmware and are upgrading to 5.3 to see if it helps. Regards Garth Darrell Long wrote: Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone877-777-2932 Direct914-271-4500 x402 Fax914-271-4292 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and firewalls?
Hi We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older Asterisk? Have there been improvements? Or is SIP (obviously depending on what client you use) still poor when it comes to NAT traversal and firewalling? Many thanks Hagen ___ --Bandwidth and Colocation provided 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 errors configuring for PRI
I moved the card to a different pci slot and that removed the error. thank you! Phil. yusuf [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 14/02/2006 15:32 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com cc Subject Re: [Asterisk-Users] Asterisk errors configuring for PRI [EMAIL PROTECTED] wrote: Hi, I've just compiled asterisk and get errors when trying to run it. Am I right in thinking that if my digium card is not plugged into our ISDN line then this is normal. If I replace zapata.conf with a blank zapata file asterisk runs fine. The reason I ask is we don't yet have ISDN installed in our new premises but I have to configure asterisk before we go in. The errors I see are: WARNING[3227]: chan_zap.c:923 zt_open: Unable to specify channel 1: Device or resource busy ERROR[3227]: chan_zap.c:6879 mkintf: Unable to open channel 1: Device or resource busy ERROR[3227]: chan_zap.c:10311 setup_zap: Unable to register channel '1-8' etc ... Thanks in advance. Phil. Hi Phil, No, the line does not have to be plugged in, the card just has to be setup correctly for asterisk to see it. after you compiled zaptel and asterisk you need to do modprobe zaptel modprobe wc (whichever card you have) ztcfg -vv this is assuming you have setup zaptel and zapata or you could do a 'make config' in zaptel folder, then restart pc, it would modprobe for you. 'lsmod' would show what is loaded. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
IMHO, the Diva Server BRI range of cards are worth considering. The Diva Server 4BRI card is an active card that can do echo cancellation, automatic gain control etc. The 4 port card costs similar to 2 single port cards so there will also be room to expand if you need it. More information can be found here: http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm It is well supported by using the Eicon Diva CAPI Driver and the Chan-capi-CM driver from Melware. Not the cheapest but you will get good quality results. I have one running here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brent Torrenga Sent: 14 February 2006 18:57 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US? We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more preferred due to it's echo cancellation abilities. I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch statement
Hi, I have two sites and I'd like to connect them with a IAX trunk and share the dialplan. Extensions cannot be clearly separated. Do I need to use 'switch' statement or DUNDI/e.164? Using 'switch', does any user can call any extension on both sites? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
My 5 cents worth is if you use Bristuff stable you must use Asterisk-1.0.10 ( Old version ) If you use Bristuff 3PRE1l you will have problems with FXO cards as I did. Bristuff3PRE1l is not Stable use at own risk!!! Regards Allan Gee Phone: +27 21 4644400 Ext. 103 www.equation.co.za -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Bagnall Sent: 15 February 2006 10:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US? I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Consensus certainly seems to be the Junghanns cards are amongst the best, but not exactly cheap. If you only need to service 2 BRIs, you might want to look at some of the passive options. We have a number of sites here in the UK running 2 HFC-S based cards in a box, all of which seem quite satisfactory (no echo, etc.). Over here you can pick up HFC-S based cards (the ones we use are these: http://www.solwise.co.uk/isdn.htm) for under £20, so they're probably even cheaper on your side of the pond. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth 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: RE : [Asterisk-Users] To connect between more than 2 asterisk server [links needed ]
Hi Thanks to all who had given advice , I had done connection between 2 IAX server , I am able to dial and communicate now , some of the problems which I faced is that when I tried to dial , it was searching was of default one. and I was getting message like Rejected connect attempt from 192.168.20.99, request '[EMAIL PROTECTED]' does not exist Feb 15 14:06:14 NOTICE[15880] chan_iax2.c: Rejected connect attempt from 192.168.20.99, request '[EMAIL PROTECTED]' does not exist I solved it by giving the required extension details on the default context , even thought this is not working according to the context, which I needed , I am happy that I am able to connect between two server Thanks to all who had helped me Thanks Joseph John --- John Joseph [EMAIL PROTECTED] wrote: --- [EMAIL PROTECTED] wrote: Hello, ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aasterisk large-scale deployment w/analog phones
hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
Hi Mr Gee I am using the Duxbury HFC PCI Bri card and found it to be very stable running asterisk-1.2.4 with Zaptel-1.2.3 with bristuff-0.3.0-PRE-1 on FedCore 4 Only problem is that you can only have FXO OR FXS on a card and not both on the same 1 port BRI card Regards Tertius Smit -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allan Gee Sent: 15 February 2006 13:01 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US? My 5 cents worth is if you use Bristuff stable you must use Asterisk-1.0.10 ( Old version ) If you use Bristuff 3PRE1l you will have problems with FXO cards as I did. Bristuff3PRE1l is not Stable use at own risk!!! Regards Allan Gee Phone: +27 21 4644400 Ext. 103 www.equation.co.za -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Bagnall Sent: 15 February 2006 10:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US? I do not even know which brands/models to consider that are out there. Given that we are in the US, and want to use BRI to improve sound quality (no echo, no static), what would be some good cards to look at? I hear a lot about BRIStuff, which I think is used on the Junghanns cards (like the quadBRI PCI ISDN), using the CAPI channel. Are those the Cadillac of ISDN cards? Consensus certainly seems to be the Junghanns cards are amongst the best, but not exactly cheap. If you only need to service 2 BRIs, you might want to look at some of the passive options. We have a number of sites here in the UK running 2 HFC-S based cards in a box, all of which seem quite satisfactory (no echo, etc.). Over here you can pick up HFC-S based cards (the ones we use are these: http://www.solwise.co.uk/isdn.htm) for under £20, so they're probably even cheaper on your side of the pond. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth 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] [Announce] Web-MeetMe v2.0.0
Hi,thanks for your quick answer.My system is Gentoo with mysql 4.1.14 installed from oficial gentoo repository. And mysql does work for other applications (I also already created the meetme db/table). Maybe the problem comes from my manual patching of the makefile to compile app_cbmysql.c (as the patch command didn't work with the makefile from 4.1.14). Compilation went fine. Dont'know. still investigating. Any help welcome.benqOn 2/15/06, Dan Austin [EMAIL PROTECTED] wrote: The error looks like a problem with the MySQL libraries on your system. I have not tested it against 1.2.4, but do have it running on SVN 7668 and have had it running on 1.2.0 I can try 1.2.4 next week if you are not able to resolve it by them. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ben QSent: Wednesday, February 15, 2006 2:46 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0 Does this work with asterisk 1.2.4?I can't make app_cbmysql work.I get an error when starting asterisk:[app_cbmysql.so]Feb 15 10:26:53 WARNING[7616]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_store_result Feb 15 10:26:53 WARNING[7616]: loader.c:554 load_modules: Loading module app_cbmysql.so failed!Who had a successfull experience compiling/running app_cbmysql.so with asterisk 1.2.4?b.en.q On 1/12/06, Dan Austin [EMAIL PROTECTED] wrote: [New Features]1.Added focus and tab-order to all input fields2.Dynamic generation of date/month/year listboxes a.It is no longer possible to schedule an invalid date. 3.Added 'Extend' and 'End Now' buttons to the monitorpage.4.Invite button on the monitor page.This greatlysimplifies the process of adding callers to a conference.The ./lib/defines file includes definitions for theprefered channel and context***5.Call history report.Support for this featurerequires the php script ./lib/cbEnd.php be running at all times.This also requires a new table in themeetme database if you're upgrading from an earlierrelease.***[Location] http://www.fitawi.com/Asterisk[Files]Web-MeetMe_v2.0.0.tgz (required)app_cbmysql.c (required)cbmysql.conf (required) cb-extensions.conf (suggested)README (suggested)[Installation]See the README[Features]1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blankii. Identify pin # conflicts (another conference withthe same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is setbut the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference3. Separate views for Current, Past and Future conferences4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added7. Users can only monitor, update or delete their conferences8. Verified administrators can monitor, update or delete anyconferences.9. Updated to Asterisk 1.2.0 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that)Thanks and enjoy,Dan***Beta testers and anyone who downloaded v2.0.0 before today The only changes from the beta was a cosmetic change to work withnon-IE browsers and a couple of installation hints.I onlyreceived feedback from one tester, so it appears the package isready to go.***Developer help/guidence request***The PHP script to monitor conference endtime andup date the CDR is fragile.If Asterisk is shutdown for more than 30 seconds, the script exits.I'd like to make it more resilent.If any PHP experts can make suggests on how to improve thescript it would be appreciated___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by
RE: [Asterisk-Users] Multiple AGI Issues
Thanks for the reply. Neat ideas there, but a couple of issues.
1. Don't want to have to jump around between the FastAGI and the dial plan. Our
plan is to have NO customer data in the dialplan, as all data will be contained
within MySQL. We don't want to have to make _any_ edits to the dial plan when a
new customer is added. It's a provisioning nightmare to have to do this. It
also may not be a Dial() command that gets excuted for a given number dialled.
It might be Meetme(), Queue() or something else. Jumping back into the dialplan
and then executing the right command would be hard to maintain. It'd be helpful
if Asterisk accepted something like the following, which would make it easier,
but it doesn't...
exten = _X.,1,AGI(//localhost/script.py)
exten = _X.,2,${APP}( ${ARGS} )
We have no customer data in the dialplan, everything is done through mysql.
There are 2 basic ways (at least) to use FastAGI when you dont want to
have a multithreaded FastAGI server.
1. Create some 'helper' contexts in the dialplan to handle
applications/actions that may take some time (like meetme,dial a.s.o.).
Let the FastAGI server set some channel variables that you may need
(like Destination number, which CallerID to use aso) and ofcourse the
context before returning to the dialplan.
2.Use a modified version of Asterisk.pm which is build around select
and nonblocking i/o that uses event driven callbacks into your
application code.(yes threadsafe, it sleeps on a select call until
events then create the callback. Since your not familiar with select i
would recommend using method 1.
What about findme/followme functionality? Are we going to have to jump
backwards and forwards between the agi and the dialplan each time (all the
while maintaining the last number tried in the agi) a new number is tried? We
could return ALL the numbers to try at once from the AGI I guess, kinda like
${NUM1}, ${NUM2}, ${NUM3} etc. Oh YUCK!
2. How did you get around the fact that it's quite clearly documented that the
perl DBI is _not_ thread safe?
You can easily use a perl distro compiled without multithread enabled
eventhough it shouldn't be needed. Again read up on perl IO::select.
You sleep waiting for event input, execute the job and then sleep
again. No multitread needed. You may issue sql request that takes 20
or 50 msec and nothing else is going on within a single server during
that time so no re-entrance into unsafe DBI code. It also means other
calls are not being served from that server during this period that why
I pointed you to use the server interleaving.
3. I don't have a high enough confidence in the stability of either perl or
python threading, to allow the FastAGI server to potentially receive several
dozen calls, and therefore several threads each. If the FastAGI server crashes,
you lost the ability to place _any_ calls.
As described above: no threading needed within the server. We have AGI
servers that processes 10K+ calls per day per server ( some of the
servers has 15K perl lines ) they never crash but they are started out
of /etc/inittab anyway (just in case)
4. Using select() system calls is a little beyond my abilities...
Doug.
I hope to get some time to do a cleanup on my framework for solution 2
above, it might benefit some other people that like to use agi-perl
b.r.
Freddi
___
--Bandwidth and Colocation provided 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 problem on 4 Processor Opteron SMP system
You can try to run "make" in the linux source-folder. I had
the same problem Running FedCore 4 on a Dual Xeon Server and running make fixed
the error
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
TeesdaleSent: 15 February 2006 12:19To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Zaptel
problem on 4 Processor Opteron SMP system
Hi All, I've just put together a system comprising of the
following; Hardware 2 x AMD Opteron 270 Processors (Dual Core)
Tyan K8WE Mobo 2GB Kingston PC3200 Registered RAM 2 x WD Raptor
1rpm 74Gb Digium TE210p Software Mandriva 2006 Public
Release (Kernel 2.6.12-12mdksmp) Asterisk 1.2.4 Zaptel 1.2.3
Problem Zaptel compiles and installs to the right place after
modifiing /usr/src/linux2.6.12-12/Makefile "EXTRAVERSION" it then installs to
"/lib/modules/2.6.12-12mdksmp/misc/". When I run "modprobe zaptel" I get
"FATAL: Error inserting zaptel (/lib/modules/2.6.12-12mdksmp/misc/zaptel.ko):
Invalid module format" uname -a reveals; "[EMAIL PROTECTED]
zaptel-1.2.3]# uname -a Linux asteriskpbx 2.6.12-12mdksmp #1 SMP Fri Sep 9
17:20:34 CEST 2005 x86_64 Dual Core AMD Opteron(tm) Processor 270 unknown
GNU/Linux" Both the kernel package and the kernel source package are
exactly the same version "2.6.12-12mdksmp and 2.6.12-12mdk"
respectively.
Regards Chris TeesdaleI.T. / I.P Telephony
DevelopmentPhilipsTel : 01325
384394 ex 246fax : 01325 383876Email : [EMAIL PROTECTED]Web : http://www.philips.org.uk
IMPORTANT: This email and any attachments may be confidential and/or
privileged. Everything is intended for use of the addressee only. If you are not
the named addressee you must not disseminate, distribute or copy this email. If
you receive this email in error please notify the sender by replying to this
email or by telephoning (+44)(0)870 609 1554 then delete this message from your
system. Philips Collection Services Ltd. ("Philips") routinely monitors the
content of email sent and received on its network, to ensure compliance with its
policies and procedures. Although Philips have taken reasonable precautions to
ensure no viruses are present in this email or any files attached to it, it
cannot accept any responsibility for any loss or damage arising from the use of
this email or its attachments and advises you to carry out appropriate virus
checks. Philips are not responsible for any changes made to the message after it
has been sent nor any files attached to it after it was sent. Emails that
contain encrypted material, program files, are obscene, inflammatory, criminal,
offensive, in breach of copyright, contain a virus or threat to computer
systems, appear to be a threat to the company or in breach of company policy may
be intercepted and/or deleted. Philips does not accept any liability for any
statements made which are clearly the sender's own and not made on behalf of
Philips.
___
--Bandwidth 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] Channel SS7
Can somebody guide me on how to get the ss7 channel up and running? I have read some information on the ss7 but I need to know which card is better and I wouldn't mind the configuration options too goksie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones
Hello, If you are doing that many analog extensions you might want to consider 4 channelbanks and a quad T1 card instead(or two 2-port cards in two servers). Four TDM24XX cards will draw a whole lot of power and would be much harder to replace than an exterior channelbank if something goes wrong with one of them. Cost should be about the same overall(depending on which channelbanks you buy), except you will be able to use a much smaller server case with the quad T1 card. The most we have done is two channelbanks off of a quad T1 card in a single machine and it works just fine. MATT--- On 2/15/06, maka [EMAIL PROTECTED] wrote: hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
On Wednesday, February 15, 2006 12:42 PM Garth van Sittert wrote: Could we possibly see your settings to get this right? I am trying to get it working at the moment. I can see the phone buttons have subscribed to asterisk, but they just don't light up. We are using 4.1 firmware and are upgrading to 5.3 to see if it helps. No problem here with bristuff and SNOM (both FW 4.x and 5.x). Have you set your hints correctly? Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk large-scale deployment w/analog phones
I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com Realize the Value of Customer Contact!TM This e-mail is intended solely for the person or entity to which it is addressed and may contain confidential or privileged material. Any duplication, dissemination, action taken in reliance upon, or other use of this information by persons or entities other than the intended recipient is prohibited and may violate applicable law. If this e-mail has been received in error, please notify the sender and delete the information from your system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Wednesday, February 15, 2006 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones Hello, If you are doing that many analog extensions you might want to consider 4 channelbanks and a quad T1 card instead(or two 2-port cards in two servers). Four TDM24XX cards will draw a whole lot of power and would be much harder to replace than an exterior channelbank if something goes wrong with one of them. Cost should be about the same overall(depending on which channelbanks you buy), except you will be able to use a much smaller server case with the quad T1 card. The most we have done is two channelbanks off of a quad T1 card in a single machine and it works just fine. MATT--- On 2/15/06, maka [EMAIL PROTECTED] wrote: hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar setuyp/planning - Do you think is it better to distribute resources among multiple (more than two), lower-port-density asterisk servers? Or is it better to use a channelbank for that purpose? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and firewalls?
Hi Hagen, It's not exactly a pleasure to run SIP through firewalls but it can be done. At least in under some circumstances. I have successfull run an Asterisk server from behind a NAT router and run a SIP trunk to the SIP VoIP provider. The problems tend to arise when multiple SIP devices wants to communicate through the NAT router. My conclusion was that all my SIP devices should be connected to the Asterisk box (ip-pbx) and that connected to the provider. To make this work I: 1. Set up port forwarding from the NAT router to the asterisk box of the relevant ports: UDP: 5004-5082 UDP: 1-2 2. Adjusted lan paramers in SIP.conf: externip = 212.xxx.xxx.xxx (I don't want your calls guys) localnet=192.168.0.0/255.255.255.0 3. Set up SIP account for VoIP provider in sip.conf . For good measure I put nat=yes in this account but I don't think it's required. 4. Make sure your SIP VoIP provider can handle NAT. 5. Extensions from outside the NAT router also works fine with this setup. Perhaps you need to set nat=yes in these extensions as well. In this configuration things are not to terrible to get going. Good luck. Cheers, John [EMAIL PROTECTED] 02/15/06 11:42 am Hi We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older Asterisk? Have there been improvements? Or is SIP (obviously depending on what client you use) still poor when it comes to NAT traversal and firewalling? Many thanks Hagen ___ --Bandwidth 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] Softphone and 911
Kyle, Right... we have hookups to Intrado at the moment and are doing it for our ATA customers. I just was trying to think if a Softphone would be compliant. Everything I've thought of seems to indicate it would be, but wanted thoughts from other people. On 2/14/06, Kyle Hagan [EMAIL PROTECTED] wrote: Softphone or hard phone doesnt matter if the service provider has the right connections to provide E911 service. We are setting up E911 compliance right now with our service. Its not as easy as just updating the address, it take time, its not instant. Kyle Matt wrote: Greetings to all, Can anyone think of a reason that a Softphone would not be compatible with the F.C.C's order for E911? If the user is able to update their address when they move their laptop, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CONFIDENTIALITY NOTICE: This message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the original message. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec issue with my iaxy
Dont know. All i know is that i had ulaw enabled in * and i was getting errors relating to iLBC. The first thing to check is whether the IAXy even does iLBC ___ --Bandwidth 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] Zap interface with Atbill
im having a problem running zap on astbill. when i dial any number through zap, astbill should minus balance if the call gets through but it minus balance even I cancle the call. any1 running astbill experienced the same ? onthe otherhand, billing on sip/iax interface is working fine, even in CDR it shows CANCLE but on ZAP interface it seems to be deducting balance on every call whether it gets through or not. --xce ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [help] warning 4246
On Tuesday 14 February 2006 19:11, fabrizio wrote: hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial(SIP/2003-bbae, zap/2/03460816149|30|t) in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel not implemented) I have a TDM400 card. with 2 channels. it seems thera are no zap channels! even in the CLI , there are no zap * commands. This indicates that chan_zap.so has not been loaded. Try load chan_zap.so from the CLi Look in modules.conf and check that chan_zap.so does not have 'noload' i.e. noload = chan_zap.so change to ;noload = chan_zap.so Lastly check if chan_zap.so exists in your modules directory... Paul -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth 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] interface to dpnss
We have 3 existing switches interconnected via dpnss, we need to integrate asterisk with these switches via a dpnss link. Any suggestions? also does anyone have a link to the differences between isdn30 and dpnss. Thanks in advance Bails ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] interface to dpnss
On Wed, Feb 15, 2006 at 01:29:10PM +, bails wrote: We have 3 existing switches interconnected via dpnss, we need to integrate asterisk with these switches via a dpnss link. Any suggestions? also does anyone have a link to the differences between isdn30 and dpnss. Get a DPNSS to something converter ... I don't think you'll find much out there to do this. I think Westell or someone make a DPNSS to H.323 box (might do SIP by now). 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 stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom buddy watch limit of 7
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If memory serves me correctly this has to do with ABE only supporting that number of watched extensions. You are correct that this is an artificial limitation and I think someone from digium actually commented that this should be improved in the future. Sean Mike Pollitt wrote: Hi All -- I've got a Polycom 601 with the sidecar unit all working with extension hints and what Polycom calls the Buddy Watch feature. I can see the state of extensions, but there seems to be a limit of 7 that I can monitor at any one time. I've put in a call to my distributor (this is how Polycom provides support). So far no response. I've seen other people have had this issue (http://voxilla.com/PNphpBB2-viewtopic-t-6350.html) but not whether anyone has successfully resolved it. Cheers, Mike. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFD8zK6y9wPyZpnL2URAhcbAJwOW8tf57pHMO8u/SHtKcUDMgNwCwCghN16 PQ9kGt3gh8wlEgyQbxxVDGg= =BFMo -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
You have to link it to the mysql libraries... add the following to the
apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c
$(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c
- -o app_cbmysql.o app_cbmysql.c
app_cbmysql.so: app_cbmysql.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB}
- -L/usr/lib/mysql -lmysqlclient -lz
Ben Q wrote:
Does this work with asterisk 1.2.4?
I can't make app_cbmysql work.
I get an error when starting asterisk: [app_cbmysql.so]Feb 15
10:26:53 WARNING[7616]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol:
mysql_store_result Feb 15 10:26:53 WARNING[7616]: loader.c:554
load_modules: Loading module app_cbmysql.so failed!
Who had a successfull experience compiling/running app_cbmysql.so
with asterisk 1.2.4?
b.en.q
On 1/12/06, *Dan Austin* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
[New Features] 1. Added focus and tab-order to all input fields 2.
Dynamic generation of date/month/year listboxes a. It is no longer
possible to schedule an invalid date. 3. Added 'Extend' and 'End
Now' buttons to the monitor page. 4. Invite button on the monitor
page. This greatly simplifies the process of adding callers to a
conference. The ./lib/defines file includes definitions for the
prefered channel and context
*** 5. Call
history report. Support for this feature requires the php script
./lib/cbEnd.php be running at all times. This also requires a new
table in the meetme database if you're upgrading from an earlier
release. ***
[Location] http://www.fitawi.com/Asterisk [Files]
Web-MeetMe_v2.0.0.tgz (required) app_cbmysql.c (required)
cbmysql.conf (required) cb-extensions.conf (suggested) README
(suggested)
[Installation] See the README
[Features] 1. Schedule new conferences a. Control start and end
times b. Set conference pin # i. Generate one if the requester
leaves it blank ii. Identify pin # conflicts (another conference
with the same pin is scheduled at the same time) c. Set Admin and
User passwords i. Generate a user password if an Admin pw is set
but the User pw is blank d. Weekly recurring conferences with the
same settings e. Select MeetMe flags per conference for Admins and
Users 2. Email the details for a successfully scheduled conference
3. Separate views for Current, Past and Future conferences 4.
Ability to modify the end time of a running conference a. Can also
reschedule a past or future conference. 5. Monitor realtime
conference activity a. Mute/Kick participants 6. Optional
authentication a. Currently Active Directory or LDAP based b.
Authentication is abstracted so unix/PAM/DB/RADIUS support could be
easily added 7. Users can only monitor, update or delete their
conferences 8. Verified administrators can monitor, update or
delete any conferences. 9. Updated to Asterisk 1.2.0 a. Changes to
the Manager interface may have caused support for 1.0.X to slip, I
cannot test that) Thanks and enjoy, Dan
***Beta testers and anyone who downloaded v2.0.0 before today
The only changes from the beta was a cosmetic change to work with
non-IE browsers and a couple of installation hints. I only
received feedback from one tester, so it appears the package is
ready to go.
***Developer help/guidence request*** The PHP script to monitor
conference endtime and up date the CDR is fragile. If Asterisk is
shut down for more than 30 seconds, the script exits. I'd like to
make it more resilent. If any PHP experts can make suggests on how
to improve the script it would be appreciated
___ --Bandwidth and
Colocation provided by Easynews.com http://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
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFD8zNjy9wPyZpnL2URAqMOAKCYVOOvHHQfNgUxVK0anBYFgCwdsgCgkJYp
I9aNkRdnmkk55GIFsjW0XwA=
=djYw
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G723 error
Hi, How do I specify a codec to use for a SIP call? IE.. If I'm doing Dial(SIP/blah) for some reason the call is connecting using the codec at the bottom of my allow list rather then top (G711u)... and I'd like to force it to G711u if possible. ___ --Bandwidth 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] which ATA SIP is better with asterisk
Hi i'm developing a solution with ASterisk, but in fact i don't know which ATA SIP device should buy. Could you give me some advices? Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
FYI... I am running this on 1.2.4 and trunk
Sean Cook wrote:
You have to link it to the mysql libraries... add the following to
the apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c $(CC) -pipe -I/usr/include/mysql
-L/usr/lib/mysql $(CFLAGS) -c -o app_cbmysql.o app_cbmysql.c
app_cbmysql.so: app_cbmysql.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK}
$ ${CYGSOLIB} -L/usr/lib/mysql -lmysqlclient -lz
Ben Q wrote:
Does this work with asterisk 1.2.4?
I can't make app_cbmysql work.
I get an error when starting asterisk: [app_cbmysql.so]Feb 15
10:26:53 WARNING[7616]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol:
mysql_store_result Feb 15 10:26:53 WARNING[7616]: loader.c:554
load_modules: Loading module app_cbmysql.so failed!
Who had a successfull experience compiling/running app_cbmysql.so
with asterisk 1.2.4?
b.en.q
On 1/12/06, *Dan Austin* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
[New Features] 1. Added focus and tab-order to all input fields
2. Dynamic generation of date/month/year listboxes a. It is no
longer possible to schedule an invalid date. 3. Added 'Extend'
and 'End Now' buttons to the monitor page. 4. Invite button on
the monitor page. This greatly simplifies the process of adding
callers to a conference. The ./lib/defines file includes
definitions for the prefered channel and context
*** 5. Call
history report. Support for this feature requires the php script
./lib/cbEnd.php be running at all times. This also requires a new
table in the meetme database if you're upgrading from an earlier
release. ***
[Location] http://www.fitawi.com/Asterisk [Files]
Web-MeetMe_v2.0.0.tgz (required) app_cbmysql.c (required)
cbmysql.conf (required) cb-extensions.conf (suggested) README
(suggested)
[Installation] See the README
[Features] 1. Schedule new conferences a. Control start and end
times b. Set conference pin # i. Generate one if the requester
leaves it blank ii. Identify pin # conflicts (another conference
with the same pin is scheduled at the same time) c. Set Admin and
User passwords i. Generate a user password if an Admin pw is set
but the User pw is blank d. Weekly recurring conferences with
the same settings e. Select MeetMe flags per conference for
Admins and Users 2. Email the details for a successfully
scheduled conference 3. Separate views for Current, Past and
Future conferences 4. Ability to modify the end time of a running
conference a. Can also reschedule a past or future conference. 5.
Monitor realtime conference activity a. Mute/Kick participants 6.
Optional authentication a. Currently Active Directory or LDAP
based b. Authentication is abstracted so unix/PAM/DB/RADIUS
support could be easily added 7. Users can only monitor, update
or delete their conferences 8. Verified administrators can
monitor, update or delete any conferences. 9. Updated to Asterisk
1.2.0 a. Changes to the Manager interface may have caused support
for 1.0.X to slip, I cannot test that) Thanks and enjoy, Dan
***Beta testers and anyone who downloaded v2.0.0 before today
The only changes from the beta was a cosmetic change to work
with non-IE browsers and a couple of installation hints. I only
received feedback from one tester, so it appears the package is
ready to go.
***Developer help/guidence request*** The PHP script to monitor
conference endtime and up date the CDR is fragile. If Asterisk is
shut down for more than 30 seconds, the script exits. I'd like
to make it more resilent. If any PHP experts can make suggests on
how to improve the script it would be appreciated
___ --Bandwidth and
Colocation provided by Easynews.com http://Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options
visit: http://lists.digium.com/mailman/listinfo/asterisk-users
--
___ --Bandwidth and
Colocation provided by Easynews.com --
Asterisk-Users mailing list To UNSUBSCRIBE or update options
visit: http://lists.digium.com/mailman/listinfo/asterisk-users
___
- --Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFD8zR9y9wPyZpnL2URAl/dAJ4hiktukLBTfvWI0Q6ubCordJDAfACeLWVM
50ffcYbq9d/ETc8JoNIWvCg=
=2/Vw
-END PGP SIGNATURE-
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
Re: [Asterisk-Users] asterisk still tries native bridging
:-((
There's nobody with any idea here? :-((.
I need to force * to not try native bridging, at least when there are
different codecs used.
In current config * tried native bridge, it fails, but CDR has been
already generated and writed :-((.
Thanks a lot for your time (and possible attention:-)
Igor
Hello,
I've problems with following -
- --- ---
PSTN | --- isdn --- | A | - iax2 -- | B |
- --- ---
On [B], there is unconditional call forwarding set back via [A]
(dialparties.agi is used) to PSTN.
So, call from PSTN is routed via [A] to [B] and than back again into
PSTN.
Everything looks good, but, after call is answered, B performs native
bridging attempt and tries to step out of voice path. And that's bad.
Because of CDR's collected from [B].
On [B] and also on [A] there is notransfer=yes in [general] section and
also in [peer/friend] definition.
It probably doesn't work. I tried to use different iax2 peer for [B]-[A]
call, so native bridging cannot occur. Fine, native bridging will fail,
but Asterisk still writes CDR.
Below is part of [B]'s config, and part of log:
[general]
bindport = 4569
bindaddr = x.x.x.x
disallow=all
allow=alaw
notransfer=yes
jitterbuffer=yes
; ---
register=sip1:[EMAIL PROTECTED] ; y.y.y.y is [A]'s ip address
; ---
[peerA]
username=sip1
type=friend
secret=Q
host=y.y.y.y
context=from-pstn
tos=0x84
notransfer=yes
jitterbuffer=yes
[peerAX]
username=sip1
type=peer ; I tried friend also
secret=QQ
host=y.y.y.y
context=from-pstn
tos=0x84
disallow=all
allow=ulaw
notransfer=yes
jitterbuffer=yes
So, incoming call comes via peerA (alaw), outgoing is made via peerAX
(ulaw).
Feb 13 15:25:48 DEBUG[27671]: Setting NAT on RTP to 4
Feb 13 15:25:48 DEBUG[27671]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Found
Feb 13 15:25:51 VERBOSE[27671]: -- IAX2/peerAX/6 is ringing
Feb 13 15:25:51 VERBOSE[27671]: --
Local/[EMAIL PROTECTED],1 is ringing
Feb 13 15:25:53 VERBOSE[27671]: -- IAX2/peerAX/6 answered
Local/[EMAIL PROTECTED],2
Feb 13 15:25:53 VERBOSE[27671]: --
Local/[EMAIL PROTECTED],1 answered IAX2/[EMAIL PROTECTED]/3
Feb 13 15:26:00 DEBUG[27671]: Planning to masquerade IAX2/peerAX/6 into
the structure of Local/[EMAIL PROTECTED],1
Feb 13 15:26:00 DEBUG[27671]: Done planning to masquerade
Local/[EMAIL PROTECTED],1 into the structure of IAX2/peerAX/6
Feb 13 15:26:00 DEBUG[27671]: Actually Masquerading IAX2/peerAX/6(6)
into the structure of Local/[EMAIL PROTECTED],1(6)
Feb 13 15:26:00 DEBUG[27671]: Got clone lock on 'IAX2/peerAX/6' at 0x8ec0ce0
Feb 13 15:26:00 DEBUG[27671]: Putting channel IAX2/peerAX/6 in 8/8 formats
Feb 13 15:26:00 DEBUG[27671]: Released clone lock on
'Local/[EMAIL PROTECTED],1ZOMBIE'
Feb 13 15:26:00 DEBUG[27671]: Done Masquerading IAX2/peerAX/6 (6)
Feb 13 15:26:00 DEBUG[27671]: Bridge stops because we're zombie or need
a soft hangup: c0=Local/[EMAIL PROTECTED],2,
c1=Local/[EMAIL PROTECTED],1ZOMBIE, flags: No,No,Yes,Yes
Feb 13 15:26:00 VERBOSE[27671]: -- Attempting native bridge of
IAX2/[EMAIL PROTECTED]/3 and IAX2/peerAX/6
Feb 13 15:26:00 VERBOSE[27671]: -- Operating with different codecs, can't
native bridge...
Feb 13 15:26:00 DEBUG[27671]: Bridge stops bridging channels
Local/[EMAIL PROTECTED],2 and
Local/[EMAIL PROTECTED],1ZOMBIE
Feb 13 15:26:00 DEBUG[27671]: Exiting with DIALSTATUS=ANSWER.
Feb 13 15:26:00 VERBOSE[27671]: == Spawn extension (macro-outsideX,
s, 6) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro
'outsideX'
Feb 13 15:26:00 VERBOSE[27671]: == Spawn extension (from-internalX,
ZZ, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2'
Feb 13 15:26:00 DEBUG[27671]: cdr_mysql: inserting a CDR record.
Feb 13 15:26:00 DEBUG[27671]: cdr_mysql: SQL command as follows:
INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2006-02-13
15:25:46','421220656111','421220656111','ZZ','from-internalX',
'Local/[EMAIL
PROTECTED],2','IAX2/peerAX/6','Dial','IAX2/peerAX/||',14,7,'ANSWERED',3,'')
Asterisk stored CDR, but call continued :-(.
Do You have any suggestion what I'm doing wrong?
I'm using Asterisk v. 1.0.9 and it's almost impossible to upgrade to 1.2.x
right now.
Thanks a lot
Igor
___
--Bandwidth 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:
[Asterisk-Users] VOIP provider iristel, setup account
I have not find any support to configure my aah 2.5 box with account from iristel Canada. I have a sip account with them and a 514 phone number assigned but I am not able to make my pbx to work with. When I call the nr from a cell Iget that nr it is not allocate. If I try to call from an asterisk extension 8 cell nr I get in log file these Feb 15 09:05:36 DEBUG[2772] manager.c: Manager received command 'Command' Feb 15 09:05:36 DEBUG[2772] manager.c: Manager received command 'Command' Feb 15 09:06:00 NOTICE[2710] chan_sip.c: Registration from 'Cristian Paun 201 ' failed for '192.168.50.155' - Username/auth name mismatch Feb 15 09:06:15 DEBUG[2710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 15 09:06:32 DEBUG[2710] chan_sip.c: Scheduled a registration timeout for irisbax.iristel.net id #1923 Feb 15 09:06:32 DEBUG[2710] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 549: Match Found Feb 15 09:06:32 DEBUG[2710] chan_sip.c: Registration successful Feb 15 09:06:32 DEBUG[2710] chan_sip.c: Cancelling timeout 1923 Feb 15 09:07:00 NOTICE[2710] chan_sip.c: Registration from 'Cristian Paun 201 ' failed for '192.168.50.155' - Username/auth name mismatch Feb 15 09:07:00 VERBOSE[2709] logger.c: -- Accepting AUTHENTICATED call from 192.168.50.145: requested format = g729, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing Macro(IAX2/206-4, dialout-trunk|2|9635279|) in new stack Feb 15 09:07:00 DEBUG[30698] pbx.c: _expression_ result is '1' Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing GotoIf(IAX2/206-4, 1?3:2)) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Goto (macro-dialout-trunk,s,3) Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing Macro(IAX2/206-4, user-callerid) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing DBget(IAX2/206-4, AMPUSER=DEVICE/206/user) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- DBget: varname=AMPUSER, family=DEVICE, key=206/user Feb 15 09:07:00 VERBOSE[30698] logger.c: -- DBget: set variable AMPUSER to 206 Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing DBget(IAX2/206-4, AMPUSERCIDNAME=AMPUSER/206/cidname) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=206/cidname Feb 15 09:07:00 VERBOSE[30698] logger.c: -- DBget: set variable AMPUSERCIDNAME to Cristian Paun Feb 15 09:07:00 DEBUG[30698] pbx.c: _expression_ result is '0' Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing GotoIf(IAX2/206-4, 0?5) in new stack Feb 15 09:07:00 DEBUG[30698] pbx.c: Not taking any branch Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing SetCallerID(IAX2/206-4, Cristian Paun 206) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing NoOp(IAX2/206-4, Using CallerID Cristian Paun 206) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing Macro(IAX2/206-4, record-enable|206|OUT) in new stack Feb 15 09:07:00 DEBUG[30698] pbx.c: Function result is '0' Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing GotoIf(IAX2/206-4, 0 0?2:4) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Goto (macro-record-enable,s,4) Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing AGI(IAX2/206-4, recordingcheck|20060215-090700|1140012420.22) in new stack Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Feb 15 09:07:01 VERBOSE[30698] logger.c: recordingcheck|20060215-090700|1140012420.22: Outbound recording not enabled Feb 15 09:07:01 VERBOSE[30698] logger.c: -- AGI Script recordingcheck completed, returning 0 Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing NoOp(IAX2/206-4, No recording needed) in new stack Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing Macro(IAX2/206-4, outbound-callerid|2) in new stack Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing DBget(IAX2/206-4, USEROUTCID=AMPUSER/206/outboundcid) in new stack Feb 15 09:07:01 VERBOSE[30698] logger.c: -- DBget: varname=USEROUTCID, family=AMPUSER, key=206/outboundcid Feb 15 09:07:01 VERBOSE[30698] logger.c: -- DBget: set variable USEROUTCID to 206 Feb 15 09:07:01 DEBUG[30698] pbx.c: _expression_ result is '0' Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing GotoIf(IAX2/206-4, 0?4) in new stack Feb 15 09:07:01 DEBUG[30698] pbx.c: Not taking any branch Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing SetCallerID(IAX2/206-4, 15149073100) in new stack Feb 15 09:07:01 DEBUG[30698] pbx.c: _expression_ result is '0' Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing GotoIf(IAX2/206-4, 0?6) in new stack Feb 15 09:07:01 DEBUG[30698] pbx.c: Not taking any branch Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing SetCallerID(IAX2/206-4, 206) in new stack Feb 15 09:07:01 VERBOSE[30698] logger.c: -- Executing NoOp(IAX2/206-4, CallerID set to 206) in new stack Feb 15 09:07:01 VERBOSE[30698] logger.c
Re: [Asterisk-Users] G723 error
I am assuming you made a profile in sip.conf like so
[sipdevice]
type=peer
host=x.x.x.x
...
.
.
disallow=all
allow=ulaw
and in extensions.conf
exten = _X.,1,Dial(SIP/sipdevice/${EXTEN})
then this MUST work. :)
you can do a sip debug or set debug 10
yusuf
Matt wrote:
Hi,
How do I specify a codec to use for a SIP call?
IE.. If I'm doing Dial(SIP/blah) for some reason the call is
connecting using the codec at the bottom of my allow list rather then
top (G711u)... and I'd like to force it to G711u if possible.
___
--Bandwidth 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] forward to gateway
hi all, hope any one can help create a trunk, i'm talking to a voip gateway provider right now, they gave me the IP address of their server a prefix to authenticate calls. How can i create such a trunk? example prefix is 1234# and IP address is 1.1.1.1, in ser i was able to do it by just simply rewriting the host part. Hope you guys can help me. TIA Regards, Ron Message sent using UebiMiau 2.7 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Announce] Web-MeetMe v2.0.0
It works!I hadn't put the rule for app_cbmysql.so: app_cbmysql.o.Not really easy to install on * 1.2.4 for non-dev people (as the patch makefile doesn't work). Thanks you very much Sean and Dan.
On 2/15/06, Sean Cook [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1You have to link it to the mysql libraries... add the following to theapps/MakefileAPPS+=app_cbmysql.soapp_cbmysql.o:app_cbmysql.c$(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c
- -o app_cbmysql.o app_cbmysql.capp_cbmysql.so: app_cbmysql.o$(CC)$(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB}- -L/usr/lib/mysql -lmysqlclient -lz
___
--Bandwidth 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] CDR for Inbound Calls
Hi What is the easiest method to set up CDRs for inbound calls? Can this be achieved without use of AGI and programming? Thanks for your help. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and firewalls?
On Wednesday, February 15, 2006 1:59 PM John Jensen wrote: Hi Hagen, It's not exactly a pleasure to run SIP through firewalls but it can be done. At least in under some circumstances. If you use a decent Firewall it will analyze and interpret the SIP Headers etc. and open the correct ports for you. Only NAT trouble left then. Again... A decent Firewall will help! Regards, JP smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Snom 360
Could we possibly see your settings to get this right? I am trying to get it working at the moment. I can see the phone buttons have subscribed to asterisk, but they just don't light up. We are using 4.1 firmware and are upgrading to 5.3 to see if it helps. Working good here in the Great White North with 1.0.9: To monitor and transfer to SIP/1000 / ext 1000: 1. Insert exten = 1000,hint,SIP/1000 into your default context (the context the extension is in) 2. In the monitoring phone's web interface, click Function Keys, pick a key, change it to Destination and type in SIP/1000. Once you submit the form it will change to a SIP URL, that's OK. 3. There is no step 3. There is a bug in the hint prioirity, this will work: exten = 1000,hint,SIP/1000 This won't work: exten = 1000,hint,sip/1000 (note the lowercase 'sip' - HAS to be uppercase) 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
Re: [Asterisk-Users] CDR for Inbound Calls
James Steven wrote: Hi What is the easiest method to set up CDRs for inbound calls? Can this be achieved without use of AGI and programming? Thanks for your help. James if I am not misunderstanding you, CDR's are automaticall written for ALL calls through the system. to specefically handle inbound calls, put it into a certain context, so you can seperate from the rest. yusuf ___ --Bandwidth 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] Fwd: Which ATA device do you recommend?
-- Forwarded message -- From: Marco Mouta [EMAIL PROTECTED] Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: [EMAIL PROTECTED] Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me their advice based on their experiencies. Thanks to all, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing a call centre app. Communication with asterisk?
Hi Arne, what you write about seems to be mostly what Flash Operator Panel does. Check it out before writing a clone yourself! :-) l. On Tue, 14 Feb 2006 13:21:37 +0100, Arne Morten Johansen [EMAIL PROTECTED] wrote: Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user is registered, busy, unavailable or available etc before she makes the transfer would be great. We have some people that are very good at programming. But for them to go on, I need to layout a plan for them on how to communicate with the Asterisk server. They have no experience with Asterisk at all, and I'm not a good programmer. My first thought is calling a PHP-script from asterisk that communicates with the java-client through IP-sockets. But I don't see how this can make the applet able to transfer calls. I'm really stuck. Anyone got suggestions and tips? Any help would be greatly appreciated. Thanks Regards, Arne Morten Johansen -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State
Hello, Currenly I've [EMAIL PROTECTED] 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: sip reason:Attack Info - Malformed SIP datagram, OPTION message is out of State By the way i've one client that is running all ok, the others all have this problem. I hope some one could help me with this. Best regards, Marco Mouta ___ --Bandwidth 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] queue_log analysis
Hi, I am running a call center based on Asterisk and building some statistics based on the queue_log file. I have some doubts about it that maybe you could help (actually, maybe these doubts are suggestions for enhancements!): 1st Scenario - Agent receives the call, and puts it on parking for somebody else to pick it up. Parking # 7000 (for attender transfer) 1140013998|1140013990.2524619|queue1|NONE|ENTERQUEUE||callerid1140014001|1140013990.2524619|queue1|Agent/5225|CONNECT|31140014016|1140013990.2524619|queue1|Agent/5225|COMPLETEAGENT|3|151140014016|1140013990.2524619|queue1|NONE|EXITWITHKEY||1 == Problems:Shouldn't a transfer to the parking extension (7000) be logged? I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially who picked up the call (whether it was picked or not), and how long did it take?What is the meaning of EXITWITHKEY in this scenario? 2nd Scenario - Agent receives the call, and transfers it to somedy else using # 1140014059|1140014051.2524641|queue1|NONE|ENTERQUEUE||callerid1140014062|1140014051.2524641|queue1|Agent/5225|CONNECT|31140014074|1140014051.2524641|queue1|Agent/5225|TRANSFER|203|default1140014074|1140014051.2524641|queue1|NONE|EXITWITHKEY||1 == Problems:I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially how long did it take?What is the meaning of EXITWITHKEY in this scenario? 3rd Scenario - Agent receives the call, and makes a blind transfer using the Transfer button of the phone (in my test, EyeBeam) 1140014104|1140014096.2524649|queue1|NONE|ENTERQUEUE||callerid1140014106|1140014096.2524649|queue1|Agent/5225|CONNECT|21140014129|1140014096.2524649|queue1|Agent/5225|TRANSFER|203|default == Problems:I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially how long did it take? 4th Scenario - Agent receives the call, and makes an attended transfer (putting the call on hold, dialing via another channel, andusing the Transfer button of the phone (in my test, EyeBeam) 1140014161|1140014153.2524663|queue1|NONE|ENTERQUEUE||callerid1140014164|1140014153.2524663|queue1|Agent/5225|CONNECT|31140014203|1140014153.2524663|queue1|Agent/5225|COMPLETEAGENT|3|39 == Problems: No transfer information is logged. Agent is considered busy (on call) until the call is actually ended, independent of the moment he actually transferred. In my agents opinion, the best way to make transfers would be the 3rd and 4th scenarios, which are obvious for phone users. But for their managers, scenarios 1 and 2 are better since more information can be used for their daily statistics. Anyway, even scenarios 1 and 2 miss lack some important statistics. Is there anybody working on enhancing this queue_log features or using any other way (maybe events and AMI) to make more complete statistic reports of call centers? Thank you very much Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Telmex PRI line configuration problem
Andres, Thanks for the explanation! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andres Enviado el: miércoles, 15 de febrero de 2006 1:31 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Telmex PRI line configuration problem no cdp enable - 1) Is it possible to have CAS framing with no R2? You cannot have CAS and PRI at the same time. Thats for sure. Also CAS goes hand-in-hand with R2. CAS means Channel Associated Signalling and MFC-R2 is the Signalling type used in each of those channels. Could be my zaptel.conf look like this? ; no cas= defined here cause the driver assigns that to MFCR2 span 1,1,0,cas,hdb3,crc4,yellow bchan=1-15 bchan=17-31 dchan=16 Some help needed Thanks in advance Oscar Andrés Carriles -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Software E.C. Along with Tellabs
Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software E.C. Along with Tellabs
Shouldn't hurt, I'd give it a try. But first you may want to fiddle with the Tellabs configuration some more. This has some good information: http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers Joseph Tanner On 2/15/06, Doug Lytle [EMAIL PROTECTED] wrote: Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BRI Newbie - What Hardware, PCI, in the US?
My 5 cents worth is if you use Bristuff stable you must use Asterisk-1.0.10 ( Old version ) If you use Bristuff 3PRE1l you will have problems with FXO cards as I did. Bristuff3PRE1l is not Stable use at own risk!!! Can't speak for anyone else, but we have 2 sites running HFC cards with Bristuff 3pre1 on 1.2.4. Both have been running 1.2 variants since before Christmas quite happily. All our ISDN devices are in TE mode, not NT mode. There may be issues with NT mode that we haven't encountered. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR for Inbound Calls
Currently, with default settings only outgoing calls are recorded. How can I enable inbound? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: 15 February 2006 15:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CDR for Inbound Calls James Steven wrote: Hi What is the easiest method to set up CDRs for inbound calls? Can this be achieved without use of AGI and programming? Thanks for your help. James if I am not misunderstanding you, CDR's are automaticall written for ALL calls through the system. to specefically handle inbound calls, put it into a certain context, so you can seperate from the rest. yusuf ___ --Bandwidth 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 unacceptable content by 'VITANIUM' the industry leading email virus and content management service from Vitanium Systems. Contact details are available at www.vitanium.com. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing a call centre app. Communication with asterisk?
quadrasoftware.com has the same app. its open source.On 2/15/06, Lenz [EMAIL PROTECTED] wrote:Hi Arne,what you write about seems to be mostly what Flash Operator Panel does. Check it out before writing a clone yourself! :-)l.On Tue, 14 Feb 2006 13:21:37 +0100, Arne Morten Johansen [EMAIL PROTECTED]wrote: Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user is registered, busy, unavailable or available etc before she makes the transfer would be great. We have some people that are very good at programming. But for them to go on, I need to layout a plan for them on how to communicate with the Asterisk server. They have no experience with Asterisk at all, and I'm not a good programmer. My first thought is calling a PHP-script from asterisk that communicates with the java-client through IP-sockets. But I don't see how this can make the applet able to transfer calls. I'm really stuck. Anyone got suggestions and tips? Any help would be greatly appreciated. Thanks Regards, Arne Morten Johansen--Loway Research - Home of QueueMetrics http://queuemetrics.loway.it___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software E.C. Along with Tellabs
Joseph Tanner wrote: Shouldn't hurt, I'd give it a try. But first you may want to fiddle with the Tellabs configuration some more. This has some good information: http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers I know, I've lived on that page during the setup of the card. I don't have a serial console to the Tellabs, nor do I have the proper wiring layout to solder a 9pin serial cable to the card, so at the moment have no options to tweak. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Software E.C. Along with Tellabs
You may want to turn the Rx gain down a bit.. -Darren -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Wednesday, February 15, 2006 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Software E.C. Along with Tellabs Shouldn't hurt, I'd give it a try. But first you may want to fiddle with the Tellabs configuration some more. This has some good information: http://www.voip- info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers Joseph Tanner On 2/15/06, Doug Lytle [EMAIL PROTECTED] wrote: Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 silence suppression?
Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail
See the problem is when I do
Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED],30)
If someone is on the phone it returns Busy and then kills the incoming
call. ChanIsAvail would work great if I was going out to the PSTN
looking for a channel, but the problem is that I need the reverse, I
need a ChanNotAvail basically saying not to ring that line.
Argh, this one has me really scratching my head.
Thanks for the info guys.
-J
I was gonna say use a queue of sorts, throw the devices into the queue
and tell it to ring all. I haven't played with it, but I would assume
that if a line's in use, it won't ring that person.
Aaron
Joseph Tanner wrote:
Perhaps I'm missing something here, but why not just have asterisk
dial all the phones regardless? No need to check what's available or
not, just dial all of them. If you don't want users on the phone to
hear a call-waiting beep, just make sure call-waiting is disabled.
Any phones that are able to ring will do so, the ones that are busy
obviously will not.
If I am missing something, let me know, but this seems to be the
easiest solution and will do what you said you need. Dial all phones,
and all that are available will ring, the rest will just return a busy
message which asterisk should ignore, as long as one phone somewhere
is not busy. I haven't run into this, but I would assume if all
phones were busy that asterisk would then go to priority +101, so you
could send them straight to voicemail.
Joseph Tanner
On 2/14/06, Jayson Navitsky [EMAIL PROTECTED] wrote:
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones that are not in use (which just so happen to be on Zap
interfaces, but might be SIP in the future).
What I have now is this (I know it's really bad):
exten = 1646555,1,Answer()
exten = 1646555,2,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],30)
exten = 1646555,3,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,4,Cut(DESK3=AVAILCHAN||1)
exten = 1646555,5,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,6,Cut(DESK4=AVAILCHAN||1)
exten = 1646555,7,Dial(${DESK3}${DESK4},30,tr)
exten = 1646555,8,Busy
(Each local is 1 zap interface)
Which is sort of my temporary work around to the problem for now,
first if there are no phones in use all phones will ring, if not it
will return busy and then it is checked to see if there is anything
available to ring between those 2 groups there. If only one phone
is in use only 2 channels will ring right now (obviously).
What I need is for any available channel to ring.
Any thoughts?
Thanks,
Jay
__
___
--Bandwidth 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] Next Montreal meeting - the 21st - featuring a conference call with Mark Spencer
Hi, This is a reminder about our next meeting. It will be held from 6pm to 8pm, February 21 at Modulis offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal. Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer. If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to select those to be presented. Please confirm your attendance at this meeting by replying to this email. See you next week, Adrien -- Adrien Laurent [EMAIL PROTECTED] www.modulis.ca (514) 284-2020 x 202 ___ --Bandwidth 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: [Amug] Next Montreal meeting - the 21st - featuring a conference call with Mark Spencer
Hi, Anybody from Québec wanting to get there with me ? I have 2 places left in my car for those who want to share the ride. Thanks, Michel Belleau SERVICES INFORMATIQUES MALAIWAH.COM (418) 261-6412 -- http://www.malaiwah.com Adrien Laurent a écrit : Hi, This is a reminder about our next meeting. It will be held from 6pm to 8pm, February 21 at Modulis offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal. Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer. If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to select those to be presented. Please confirm your attendance at this meeting by replying to this email. See you next week, Adrien -- Adrien Laurent [EMAIL PROTECTED] www.modulis.ca (514) 284-2020 x 202 ___ Amug mailing list [EMAIL PROTECTED] http://lists.modulis.ca/mailman/listinfo/amug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Snom 360
Garth, Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3. Reagrds - Original Message - From: Garth van Sittert [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 15, 2006 12:41 PM Subject: Re: [Asterisk-Users] Asterisk and Snom 360 Could we possibly see your settings to get this right? I am trying to get it working at the moment. I can see the phone buttons have subscribed to asterisk, but they just don't light up. We are using 4.1 firmware and are upgrading to 5.3 to see if it helps. Regards Garth Darrell Long wrote: Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone877-777-2932 Direct914-271-4500 x402 Fax914-271-4292 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth 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] Newbie question
Hi there,I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive.I dont now which card to take.Please tell me what you think about. I appreciate all suggestions.Thanks in advanceHousi Mueller Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel SS7
Have some NMS TX4000-4link Full stack for sale. Mark www.voiceinternational.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)
Nik, Looks like you're making some progress. When I first started using [EMAIL PROTECTED] I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten = 555,1,Dial(Zap/1/5595551212) I chose a specific Zap channel and the exact digits that I wanted to send to the telephone company. This helped me figure out what to dial. The other thing you can do is log on to the CLI and turn on PRI debugging: pri debug span 1 This will cause PRI debug messages to display on the console. It might take a while but you will learn to read those debug messages. You can also post them to the list and we'll help you to interpret them. -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nik600 Sent: Wednesday, February 15, 2006 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion) On 2/14/06, Michael Collins [EMAIL PROTECTED] wrote: Nik, I'm not sure that NOP is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have OK under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. ok, with your configuration incoming calls works, but: - i have eco (maybe i have to increase/decrease echotraining value?) - outgoing calls doesn't works ( -- Executing Dial(SIP/102-cc9b, ZAP/g0/mynumber) in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto(SIP/102-cc9b, s-CONGESTION|1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp(SIP/102-cc9b, Dial failed due to CONGESTION) in new stack -- Executing Macro(SIP/102-cc9b, outisbusy) in new stack -- Executing Playback(SIP/102-cc9b, all-circuits-busy-now) in new stack ) it seems that i don't have any channel for outbound - ALARM is set on NOP i've got a TE205P and my zaptel.conf is: span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone= it defaultzone = 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
Re: [Asterisk-Users] Newbie question
On Wed, 15 Feb 2006 08:59:22 -0800 (PST) housi mueller [EMAIL PROTECTED] wrote: Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you think about. I appreciate all suggestions. Thanks in advance Housi Mueller My personal preference would be to go with the E1/T1 now. It would give you expansion opportunities in the future between the Asterisk and the Panasonic, allow you to be all digital between, and finally if you ever decided to ever get rid of the Panasonic, you could pull a T1 from the telco straight into the Asterisk box. Spend a little more now and save in the future. Just my $.02 Robert ___ --Bandwidth and Colocation provided 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 large-scale deployment w/analog phones
Hunt, Bill wrote: I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point This list is not for advertising in any form (that's what 'non-commercial discussion' means). The cards the original poster mentioned also do on-card bridging of channels (in fact, as far I'm aware pretty much all the current Asterisk compatible interface cards do it). It's not a competitive advantage for anyone :-) ___ --Bandwidth 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] arris e-mta
Hi This may be off topic because it involve cable. I am testing with Arris cable modem / MTA I have 2 models, one older and one newer. With older one, everything works fine With the new one, I can register, make a call and I hear the other person but he can't hear me The config is the same with both units except for the username of course, Anybody ever worked with these units ? Our CMTS is a uBR7246VXR from Cisco Thanks Patrick ___ --Bandwidth 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] Channel bleedover?
Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? Thanks! Paul ___ --Bandwidth 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: Channel bleedover?
I've had pretty good luck getting the telco to bring out a laptop and test the lines for this sort of thing. Not past the DMARC, of course, but still it helps to narrow problems down. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul A. Pringle Sent: Wednesday, February 15, 2006 11:50 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Channel bleedover? Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? Thanks! Paul ___ --Bandwidth 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] Hint priority
Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call! In sip.conf I have: [11] callerid=Reception 11 username=11 secret=XXX type=friend host=dynamic dtmfmode=rfc2833 mailbox=11 context=internal subscribecontext=internal [12] callerid=John 12 username=12 secret=XXX type=friend host=dynamic dtmfmode=rfc2833 mailbox=12 context=internal subscribecontext=internal And extensions.conf [internal] exten = 11,hint,SIP/11 exten = 11,1,Macro(dial-extension,11) exten = 12,hint,SIP/12 exten = 12,1,Macro(dial-extension,12) A 'sip show subscriptions' gives: Peer UserCall ID ExtensionLast state Type 10.0.0.1011 3c267009cf8 11 Idle dialog-info+xml 10.0.0.1011 3c267009b71 12 Idle dialog-info+xml Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bleedover?
On Wednesday 15 February 2006 12:49, Paul A. Pringle wrote: Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? I imagine you're talking about a TDM2400 here (you don't specifically mention, but 11 lines suggests that is the case) -- I would first check your gains and wiring; the TDM card shouldn't have any design issues which would cause crosstalk like this, and once the information is digitized it's impossible to get this effect unintentionally. However it is trivial to have high gains (or improper attenuation) cause crosstalk on analog circuits, especially poorly wired cross-connects. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-941 stutter tone
I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bleedover?
Paul A. Pringle wrote: Occassionally on calls we get what sounds like low volume channel bleedover. Not clear enough to make out words, but not echo of either side of the main coversation. We're using a Digium card with 11 channels connected to PSTN lines. Any ideas on what the problem is or how to go about troubleshooting? A Digium card? Can you be slightly more specific? Generally speaking, you should contact Digium Support when you have issues with Digium hardware, as they are best equipped to deal with these issues. ___ --Bandwidth and Colocation provided 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 silence suppression?
Asterisk DOES NOT HAVE silence suppression (VAD) support for now. So it cannot be disabled or enabled. Simply does not exists. A couple of weeks ago i saw a patch to enable it. The link here: http://bugs.digium.com/view.php?id=5374 so unless you have the previous patch, you should disable silence suppression in the clients. Regards On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads?Thx as always___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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
[Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX
Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA’s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software. “The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA’s DSP processing power on the board,” stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. “Because of the native bridging for TDM calls, latency is drastically reduced—nearly eliminated— in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments.” Stroudwater Contact Point, LLC, based in Portland Maine, provides software development services, applications and infrastructure for contact centers. We have chosen PIKA's Daytona MM analog hardware for analog support in their upcoming Asterisk-based Dirigo iQueue™ PBX/ACD product because of the scalability and density of 24 ports for either loop start or POTS. On board switching of calls and on board echo cancellation make the product more efficient and ease demands on the server's CPU. Further, PIKA support throughout our development has been outstanding, said Bill Hunt, President and CTO at Stroudwater Contact Point. Our next step is to integrate the PIKA on board fax solution. Unlimitel Inc. offers VoIP services to business customers across Canada using the VoIP/PSTN network and was looking for a way to deliver reliable fax. Stephan Monette, President of Unlimitel stated: “Using Asterisk and the PIKA [high density analog] Daytona board was quick and easy. We were able to demonstrate the stability of the fax service in our lab within 24 hours!” Kanatek Technologies Inc. is an Ottawa, Ontario-based systems integrator delivering IT consulting and support services to a variety of companies in North America. Paul Labelle, Vice President of Operations at Kanatek said, “We were impressed with the flexibility and customization that the PIKA Connect for Asterisk solution could provide. We were able to integrate it with our current PBX system, and extend our communications network to nearly 100 users at multiple locations including the corporate office, branch office and remote locations.” Asterisk developers can be up and running quickly with PIKA Connect for Asterisk and PIKA hardware. “For those familiar with using the Asterisk platform, no additional training is required. They can take advantage of the PIKA solution with minimal effort or investment,” said PIKA Technologies’ Wojciech Tryc. For more information on PIKA Connect for Asterisk, go to http://www.pikatechnologies.com/products/asterisk.htm About PIKA PIKA Technologies designs and manufactures computer plug in voice cards and software that connect a computer system to both TDM- and IP-based networks to provide advanced voice services. For almost two decades PIKA Technologies has been serving companies around the world that require voice cards to design sophisticated phone services for recording systems, voice services applications, and PC-PBX systems. The company has built a reputation for delivering innovative products and exceptional technical support by working closely with its customers. Headquartered in Ottawa, ON, Canada, the company has ranked in The Branham300, an authoritative ranking of successful Canadian high tech firms, for three consecutive years. Visit www.pikatechnologies.com or call +1-613-591-1555 for more information. © PIKA Technologies Inc., 2005. PIKA is a registered trademark of PIKA Technologies Inc. For more information, please contact: Miriam Rautiainen Head of Marketing | PIKA Technologies Inc. ___ --Bandwidth 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] Automated wake up call
Does anyone have any system in place that does automated wake up calls. With recordings and options configurable over the phone? -- Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: ZAP extension, DTMF?
Hi Dan, How is your echo can the issue? Did you disable the echo can and solve the DTMF issue? If you did, did it trade the DTMF issue with echo problem? It would nice if you can share your experience. Thanks. Andy On 2/14/06, Dan Elder [EMAIL PROTECTED] wrote: Please ignore my last query about DTMF on ZAP, turned out to be an echo can issue. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided 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 silence suppression?
The patch you saw is not for the stable branch. Salu2 Jsalas -Mensaje original-De: Moises Silva [mailto:[EMAIL PROTECTED]Enviado el: Wednesday, February 15, 2006 2:28 PMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] asterisk silence suppression?Asterisk DOES NOT HAVE silence suppression (VAD) support for now. So it cannot be disabled or enabled. Simply does not exists. A couple of weeks ago i saw a patch to enable it. The link here:http://bugs.digium.com/view.php?id=5374so unless you have the previous patch, you should disable silence suppression in the clients.Regards On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads?Thx as always___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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] Developing a call centre app. Communication with asterisk?
Hello, The astGUIclient web-client does most of this, it is open source and entirely web-based so no need for JAVA: http://astguiclient.sourceforge.net/ MATT--- On 2/14/06, Arne Morten Johansen [EMAIL PROTECTED] wrote: Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user is registered, busy, unavailable or available etc before she makes the transfer would be great. We have some people that are very good at programming. But for them to go on, I need to layout a plan for them on how to communicate with the Asterisk server. They have no experience with Asterisk at all, and I'm not a good programmer. My first thought is calling a PHP-script from asterisk that communicates with the java-client through IP-sockets. But I don't see how this can make the applet able to transfer calls. I'm really stuck. Anyone got suggestions and tips? Any help would be greatly appreciated. Thanks Regards, Arne Morten Johansen ___ --Bandwidth 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] Alarmreceiver
Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under linux? Andrutto Cheers --- Fotoerotica! http://link.interia.pl/f1904 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX
Wojciech Tryc wrote: Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today announced Take this to the -biz list... This is for asterisk discussion, not marketing. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alarmreceiver
Quoting andrutto [EMAIL PROTECTED]: I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I don't see anywhere in the wiki where it says this is unreliable. The wiki mentions that This application is NOT Underwriter's Laboratory (UL) approved. My experiance is that it is as reliable as anything else in Asterisk. I've been using it since August of 2004 and it's always worked fine for me. I use a DMP security system with card access. When somone opens a door to my house with their card, it reports the event to Asterisk which then annoucnes the name of the person who as come in (and what door)throughout the house. I also have a text message sent to my pager of the event. I've also used it with a GE panel to send alarms from a remote Central Office. I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Can you be more specific? How is the alarm panel connected to the Asterisk system (ATA, ZAP Channel, etc) This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [CAVPdiscussion] OT: RFC: Canadian Association o f Voice over IP Users (CAVU)
In the latest CAVP conference call, the membership body voted to restrict membership to VoIP LEC's and to create a seperate membership body for any other parties interested in contributing to the CAVP's efforts in CRTC lobbying and providing a unified industry presence in the Canadian telco industry with a view to VoIP services. Accordingly, I would like to propose an adjutant association to CAVP: the Canadian Association of Voice over IP Users. CAVU is proposed to be a membership body that Voice over IP users can join and advise CAVP on issues common to CAVP and CAVU. Since the majority of CAVP members are also Asterisk users, CAVU is also proposed to be also an Asterisk-specific user group. Membership would be open to any entity or individual that: -Is in Canada -Uses Asterisk or a Voice over IP platform or product in a meaningful way -Is interested in Voice over IP regulation and interop between the PSTN, ILEC's, CLEC's, the CAVP membership body, private businesses and users -Is interested in creating Asterisk-specific freely avaliable solutions that allow implementors to create installations that conform to future CRTC regulations and specific challenges that Canadian VoIP users face. Ongoing topic, activity, or working group suggestions may include: -Asterisk ENUM compliance according to the CIRA recommendations -A working group to define an RPC protocol for an IP enabled 911 PSAP -Canadian - specific implementation details for Asterisk connectivity to Canadian telco and cell networks -Hacking on Asterisk for neat-o applications (hooking Canada411.com, for example) -Meets where possible, with interesting demos Because of the wide geographic nature of Canada, sometimes a physical F2F user group meeting would not be possible. CAVU would address this with conference bridges and a PHP-BB type of web based system and / or a mailing list. Of course, if there's interest enough in a particular geographic region, CAVU could form regular or non-regular meets to achive our common goals and maybe share a few laughs and war stories. If anyone's in Edmonton, for example, let's go down to Whyte for a few beers. It is proposed that this association be a loose, ad-hoc association without too much formal pomp and circumstance. It is further proposed that meaningful work or recommendations produced by CAVU be forwarded to CAVP for presentation to regulatory bodies in a formal fashion. As this is a testing the waters phase, I'm just gaguing interest. If you are interested in participating, or have a positive or negative comment, one way or another, ping me an email at [EMAIL PROTECTED] - who you are, what your take on this is, and where you are. If there are enough people to form a group, we can form the (loose) association and begin working with CAVP in order to define some priorities and flesh out the mandate of the association. Or we can hack on Asterisk. Whatever. More information on the CAVP: http://www.cavp.ca CAVU contact email: [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] asterisk silence suppression?
The silence suppression is a client setting. Asterisk does not have silence suppression as far as I know. Garth Dan Elder wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX
On Wed, Feb 15, 2006 at 01:28:39PM -0500, Wojciech Tryc wrote: Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA’s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software. So where can I get my hands on those drivers? [ snip strange marketing stuff ] For more information on PIKA Connect for Asterisk, go to http://www.pikatechnologies.com/products/asterisk.htm /me follows link /me gets stopped by the guard at http://www.pikatechnologies.com/extranet/portal.htm asking for a username pssword Must I really register to get those GPLed drivers? Any kind soul willing to mirror them? (they are freely distributable, right?) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and firewalls?
Well, netfilter is a decent firewall :). Give the sip-conntrack helper a try, and then please tell me what u found. see: www.iptel.org/sipalg for help. Cheers. Mensaje citado por: \\\Koopmann, Jan-Peter\\\ [EMAIL PROTECTED]: On Wednesday, February 15, 2006 1:59 PM John Jensen wrote: Hi Hagen, It\\\'s not exactly a pleasure to run SIP through firewalls but it can be done. At least in under some circumstances. If you use a decent Firewall it will analyze and interpret the SIP Headers etc. and open the correct ports for you. Only NAT trouble left then. Again... A decent Firewall will help! Regards, JP __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP and firewalls?
Well, netfilter is a decent firewall :). Give the sip-conntrack helper a try, and then please tell me what u found. see: www.iptel.org/sipalg for help. Cheers. Mensaje citado por: \Koopmann, Jan-Peter\ [EMAIL PROTECTED]: On Wednesday, February 15, 2006 1:59 PM John Jensen wrote: Hi Hagen, It\'s not exactly a pleasure to run SIP through firewalls but it can be done. At least in under some circumstances. If you use a decent Firewall it will analyze and interpret the SIP Headers etc. and open the correct ports for you. Only NAT trouble left then. Again... A decent Firewall will help! Regards, JP __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. ___ --Bandwidth 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] Bridge Calls with G()
Hi Guys,
This article was posted few days back. I thought i can get more info here.
I am trying to bridge two outbound calls together. (have a program start a
context, dial one party and then bridge another party)
I thought that the G() flag in the dial application would work.
I tried the the following test (continue down a dial plan). One station
calls into a context ... in this case, dials '55' to start the extenion
exten = 55,1,DBget(PHONE2=demo/phone2)
exten = 55,2,Playback(/recordings/prompt01)
exten = 55,3,Dial(${PHONE2},,rG(from-internal-custom, 55, 4))
exten = 55,4,Playback(/recordings/prompt02)
exten = 55,5,Hangup()
You would think that the two parties would hear prompt02 and each other in a
conversation ... this does not work. However:
exten = 55,1,DBget(PHONE2=demo/phone2)
exten = 55,2,Playback(/recordings/prompt01)
exten = 55,3,Dial(${PHONE2},,rG(from-internal-custom,55,4))
exten = 55,4,Playback(/recordings/prompt02)
exten = 55,5,Playback(/recordings/prompt04)
exten = 55,105,Hangup()
This works but the calling party hears 'prompt02' and the called party hears
'prompt04' the two parties are NOT connected foa conversatoin - just like
the wiki describes
Does anyone know when the 'G()' flag will be fixed or any potential
work-arounds?
Thanks,
Prakash
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
