[Asterisk-Users] Sangoma card problem with EWSD Exchange
Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange,connection with E1, CAS, (using unicall-0.0.3pre4). my systemrun success,incoming call and call out are good. when iswitch to EWSD (SIEMENS) R-15. my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10 (successfull) Aug 17 13:25:38 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 0001 [1/ 1/Idle /Idle ]Aug 17 13:25:38 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 DetectedAug 17 13:25:38 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 Making a new call with CRN 32769Aug 17 13:25:38 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1101 - [2/ 2/Idle /Idle ]Aug 17 13:25:38 WARNING[21952] chan_unicall.c: Unicall/15 event DetectedAug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 0 on [2/ 2/Seize ack / Seize ack ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 on - [2/ 2/Seize ack /Seize ack ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 0 off [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 off - [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 0 on [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 on - [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 0 off [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 off - [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 2 on [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 on - [2/ 2/Group A /DNIS req uest ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 2 off [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 off - [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 2 on [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 on - [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 2 off [2/ 2/Group A /DNIS request ]Aug 17 13:25:39 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 1 off - [2/ 2/Group A /DNIS request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - E on [2/ 2/Group A /DNIS request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 on - [2/ 2/Group A /DNIS request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - E off [2/ 2/Group A /Category req ]Aug 17 13:25:40 WARNING[21 952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /Category req ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 on [2/ 2/Group A /Category req ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 on - [2/ 2/Group A /Category req ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 off [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 7 on [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 on - [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 7 off [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MF C/R2 UniCall/15 - 1 on [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 on - [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 off [2/ 2/Group A /ANI request ]Aug 17 13:25:40 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 8 on [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 on - [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 8 off [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 on [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MF C/R2 UniCall/15 5 on - [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 off [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 5 off - [2/ 2/Group A /ANI request ]Aug 17 13:25:41 WARNING[21952] chan_unicall.c: MFC/R2 UniCall/15 - 1 on [2/ 2/Group A /ANI request ]Aug 17 13:25:41
Re: [Asterisk-Users] /etc/init.d/asterisk barfing
Paul Belanger wrote: #root service asterisk start Starting asterisk: [ OK ] # ps aux does asterisk show up as a process? nope. But it does if I manually type safe_asterisk or asterisk Julian PB ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pleiades p32mxi
HI, Has someone sucessfully connected this channelbank to an asterisk through a digium card? I would rather use this channel bank, since is the only one I have! thanks to all Alejandro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma card problem with EWSD Exchange
Nguyen Trung Tin wrote: Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD exchange store in two provinces difference. in Vietnam) this is my logfile of E10 (successfull) Have you tried contacting Sangoma? -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Queues and Strategies
--- Waldo Rubinstein [EMAIL PROTECTED] wrote: I tried the same experiment with all the queueing strategies and the behavior was the same. The only exception was with ringall. The problem with ringall is that it shows the same caller-ID to all agents. Once the first agent picks up and the next call rings in all remaining phones, the caller-ID is now reflecting the caller-ID of the new call, but that of the old call (may be a bug). you mean to say that the new caller-ID is not reflection it was the old caller-ID only is it so ? __ Yahoo! India Matrimony: Find your partner online. Go to http://yahoo.shaadi.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommendation for 8 lines analog card in Australia
Hi, we want to build a Asterisk server for a branch office in Australia. At the moment they use 5 analog lines. We will need at least 8 lines. What hardware would you recommend for the 8 analog PSTN lines? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk run problem, was working, rebooted server, now nothing
I've copied over all the asterisk configuration settings. There was nothing decent to see in the logs, so I#12288;didn't copy those. http://zanshin.tsumelabs.com/ The system was working a couple days ago until I had the server rebooted. there are 2 zap cards, both are working fine, all lines are able to recieve phone calls. The problem is callers call the building and they are intruduced to the weekday message, then when they hit 1 they are placed in a queue for appointments. They are always on hold and the receptionist phone doesn't ring. What could be the problem? receptionist == 101 Tsume ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED]: How to changed AMP User Login andPassword
From the command prompt type: help-aah This will give you a list of commands to change passwords. For example: Commands Descriptions --- config set the local time zone and keyboard type netconfig configure ethernet interface genzaptelconf autoconfig Zaptel cards restore-aah restore from a backup install-AVMB1ISDN install support for AVB B1 ISDN card install-EiconDiva install support for Eicon Diva ISDN card install-pdf installs support for emailing PDFs of faxes passwd-maint set master password for web GUI passwd-amp set password for amp only passwd-meetme set password for Web MeetMe only passwd set root password for console login passwd admin set admin password for checking system mail setup-cisco create a SIPDefault.cnf in /tftpboot setup-dhcp set up a dhcp server rebuild_zaptel rebuild zaptel driver after kernel update asterisk -r Asterisk CLI yum -y update Get latest patches for CentOS From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Zeeshan Zakaria Sent: Wednesday, August 31, 2005 7:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED]: How to changed AMP User Login andPassword Hi, I cant figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE cards with ISDN BRI?
Does anybody know if the Digium TE series cards will work with NI-1 (SBC California) ISDN BRI? If not can anyone make recommendations as to reliable cards to use? My end goal is to use the BRI lines for incoming fax (spandsp) only. Thanks in advance, Aaron Picht ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE cards with ISDN BRI?
On Thu, 1 Sep 2005, Aaron Picht wrote: Does anybody know if the Digium TE series cards will work with NI-1 (SBC California) ISDN BRI? If not can anyone make recommendations as to reliable cards to use? My end goal is to use the BRI lines for incoming fax (spandsp) only. I cannot tell you anything about the digium cards, but the Eicon DIVA Server cards do support NI-1 and using the onboard DSPs, you can fax without spandsp. Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /etc/init.d/asterisk barfing
Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to require a keypress on answer?
[apologies if this comes through twice - the original doesn't seem to have shown up even after 16 hours] In the handling of agents, when using AgentCallbackLogin, a call placed to an agent needs to be accepted by the agent pressing the '#' key. I'm trying to replicate that kind of operation in a non-agent scenario: I want to call Dial() from my dialplan, play an announcement to the called party if they answer, and then for the dialplan to be able to tell if the called party pressed a key or not. This is for an alarm application to know it got through to a person and not just an answering machine. I've tried using the M() option to call a macro, but any channel variables created in the macro are created in the called channel and not the calling channel, so I can't use them to pass status back to the dialplan. This is on a system running Asterisk 1.0. Any ideas gratefully received! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /etc/init.d/asterisk barfing
Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Hello Paul, Thursday, September 1, 2005, 4:38:42 AM, you wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) Same here, it think it depends on hint status: when you make a call calling hint is set to 1, but called one stays 0. Correct behaviour should be put the hint of the caller to 1 (steady ligt) while calling put the hint of the called to X (blinking light, cant remember which state it is ) while phone is ringing, then to 1 if call is answered. Unfortunately I dont know how to accomplish this Regards! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mulig_SPAM: More than one outgoing call
Hi I have setup a queue with 2 agents in it... one on an extension the other an outgoing call - Cell phone If I have to callers in the queue, and pickup the the first caller with my cell phone the other caller gets a all circuits are busy, please try again later Why is that. I have setup my trunks up, so they can handle 10 calls inbound/outbound. And in my outbound routing I use 2 trunks... Because I have two phonenumbers. What could be the solution to it. /Tue ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P problems
Hello, We recently bought 2 TDM400P Rev I boards with in total 8 FXS ports to be used with Asterisk. I use Asterisk CVS Head (version around 15 aug 2005). I have an ISDN Quad boards towards the national Telco. The TDM400P has his own Interrupt line. I encounter 2 major problems: 1) Transmitting and receiving van Fax is very unreliable (on the CLI I see a Native bridging (seems to be 911.ulaw, 64 kbit high quality). Sometimes the Fax is Ok, sometimes I miss a few lines, sometimes it's impossible. The same problem is there when sending or receiving faxes. 2) I'm trying 3 different Analog phones and having 3 different behaviors: 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri ring... ri...) 1 phone does not ring at all when Asterisk says 'Ringing Zap/6'. However, when I do an 'off-hook' on this phone, I get a normal tone signal and I can dial and talk perfectly. DTMF is recognised too. It simply does not ring on incoming calls. The 3 phones are compliant to the Belgium Telco regulations and work perfectly in a Telco POTS line. Are there differences in 'Ring Voltage' ? Can I twist some settings to solve these problems ? Thank you -- Alex Ongena Managing Director --- Able N.V.Tel: +32(0)15 50.44.00 Dellingstraat 28bFax: +32(0)15.50.44.09 B-2800 Mechelen Belgium mailto:[EMAIL PROTECTED] http://www.axsguard.com http://www.doITsafe.net aXs GUARD - internet communication appliance --- -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why ZAPATA inserting pause before last digit, during dialing? GRRRRR....
I want to speed-up dialing on X101P clone (Ambient modem). I probably must change wcfxo.c, but what line to change? I found what to change: digits.h line 23 from #define DEFAULT_DTMF_LENGTH 100 * 8 to #define DEFAULT_DTMF_LENGTH 50 * 8 and my dialling is now much faster. But, I have new question: Before last digit, there is always inserted pause (500ms) or maybe two (1000ms). I don't use pause anywhere in my dial-plan, so, why is inserted and dialled? To test is that really a pause or something else, I changed line 25 from #define PAUSE_LENGTH 500 * 8 to #define PAUSE_LENGTH 2000 * 8 and, guess what, now I have 4sec pause before last digit is played. How to get rid of it? I can maybe #define PAUSE_LENGTH 0 * 8 but that this is very dirty solution. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange problem with Bristuff
Hi all, I have a strange problem with a quadbri card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed. I have connected to the card 3 isdn in ptp mode configured in selection passing (I don't know if is exact the english traduction but I have 3 isdn with 99 numbers and asterisk forward the extensions) The problem is this: if I call from a cellular to asterisk all is Ok but when I try to call from a fixed line the extension (the last part of the number) is not sent but only the first part. Someone can help me ? Bye___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to execute StopPlayTones when a SIP phone is answered
I'm trying to find a way to generate an 'internal
extensions' tonelist but I can't seem to find anything on how to do this.
My idea was to start a Playtones(intercom) tonelist and not indicate ringing to
the line (dead air). But then, somehow StopPlayTones needs to be run once
the ringing telephone picks up.
This seems like a dirty way to do this. I
envision an option to the Dial cmd's option 'r', where you could specify a
ringtone to play if not the default, i.e.
In indications.conf:
[us]
...
...
ring =
400+450/400,0/200,400+450/400,0/2000intercom =
400+450/400,0/200,400+450/400,0/2000 ;FRESHLY ADDED AND STOLEN FROM [uk]
section.
1001,1,Dial(SIP/1001,20,r{intercom})
For what its worth, I'm trying to use the standard
UK ringtones for an internal extension. This behavior mimics several
different PBXs and KSUs on the market.
Does anyone have something like this
working?
Chris Coulthurst
[EMAIL PROTECTED]
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[Asterisk-Users] Help setting up trunk on AAH
Hi everybody, Ive proxy server IP, user ID and password. Now I need to connect to a remote Asterisk server as a SIP using my Asterisk @ Home box. That Asterisk server will make PSTN calls for me. I think I am making mistake while setting up the Trunk because when trying to make calls, it give all circuits are busy error. When I setup Sipura adapter, which is relatively easier to setup, everything goes smooth. I have following configuration in AAH: Outbound Caller ID: My name and number Maximum channels: 1 Dial Rules: empty Outbound Dial Prefix: empty Trunk Name: Outbound Trunk Peer Details: host=IP of the other asterisk secret= type=friend username=1096773 User Context: empty User Details: empty Registration String: [EMAIL PROTECTED] of other asterisk server:password Please tell me what I need to do so that I can make calls from my AAH this trunk asterisk server Thanks Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Issue #3644 has recently been committed to CVS-HEAD which allows for full device state notification via subscriptions for Snom 360 and other supporting phones w/o the need for additional patches. On 9/1/05, Alessio Focardi [EMAIL PROTECTED] wrote: Hello Paul,Thursday, September 1, 2005, 4:38:42 AM, you wrote:PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming callPH the light does not come on.PH I do not want to install the bristuff patch if possible.PH (although I can see that with the devstate command I can make the lights PH do whatever I want)Same here, it think it depends on hint status: when you make a callcalling hint is set to 1, but called one stays 0.Correct behaviour should beput the hint of the caller to 1 (steady ligt) while calling put the hint of the called to X (blinking light, cant remember whichstate it is ) while phone is ringing, then to 1 if call is answered.Unfortunately I dont know how to accomplish this Regards! --Best regards,Alessiomailto:[EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P problems
On Thursday 01 September 2005 06:35, Alex Ongena wrote: I encounter 2 major problems: 1) Transmitting and receiving van Fax is very unreliable (on the CLI I see a Native bridging (seems to be 911.ulaw, 64 kbit high quality). Sometimes the Fax is Ok, sometimes I miss a few lines, sometimes it's impossible. The same problem is there when sending or receiving faxes. This is a well-known issue with the current drivers, although nobody's really stepped up to identify when exactly this started happenning (as the driver code was changed) or why. 2) I'm trying 3 different Analog phones and having 3 different behaviors: 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri ring... ri...) 1 phone does not ring at all when Asterisk says 'Ringing Zap/6'. However, when I do an 'off-hook' on this phone, I get a normal tone signal and I can dial and talk perfectly. DTMF is recognised too. It simply does not ring on incoming calls. Have you tried the 'boostringer=1' module option? If you swap phones and ports around (i.e. try phone #3 in phone #1's port) does the problem stay with the phone or the port? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange problem with Bristuff
On Tue, Aug 30, 2005 at 07:45:50PM +0200, [EMAIL PROTECTED] wrote: I have a strange problem with a quadbri card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed. I have connected to the card 3 isdn in ptp mode configured in selection passing (I don't know if is exact the english traduction but I have 3 isdn with 99 numbers and asterisk forward the extensions) The problem is this: if I call from a cellular to asterisk all is Ok but when I try to call from a fixed line the extension (the last part of the number) is not sent but only the first part. It might be a pattern matching problem in your dial plan. The digits used as extension (0,..,9,00,...,99) might be transmitted one by one or as block. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 1001 Adapter with two lines using one RG11 jack
Hi, Ive Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. Ive [EMAIL PROTECTED] server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing something wrong or this adapter doesnt support this feature of two separate lines? Thanks Zeeshan A Zakaria ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What this little red cross mean in AAH
Hi, In asterisk at home, in Outbound Routing menu, under the trunk sequence (e.g. IAX2/FWD), what does little red cross mean beside the selected trunk. Thanks Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: equipment configuration help
Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have to buy an analog card? or maybe link me to a web site that explains (with images) how a t1/e1 is managed? Thanks, and I apologize for this completely newbie question. I've never seen images or instructions on how to handle this. Im not even sure im using the right terms in Google. -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To
Re: [Asterisk-Users] Snom 360 and hints
PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) First, ensure that the 360 has Filter Packets from Registrar turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do sip show subscriptions from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone vmail indicator and TDM400P woes
2) I have 2 TDM400Ps installed in a system. I need the audio on the analog phone (FXS modules) to be amplified somewhere between 10 and 15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS interfaces. When a call comes in on the FXO at this setting, the call sometimes has about 20 seconds of loud noise that sounds like it might be music in the background. The problem is intermittent. Any ideas on how to solve this one? What settings should I use so that everything is at default gain (0.0) except the volume on the analog phone sounds louder? I'm guessing txgain=10.0 (or whatever value) on the FXS module? Just a couple of comments on setting the gain controls You don't mention why you might need 10 or 15 db gain, but keep the following in mind. - the gain settings for any fxo pstn connections are intended to compensate for the loss associated with that connection, and should be set to allow pstn calls to hit your asterisk box as close as possible to 0 db. (0db is not realistic for analog pstn links via a TDM card with fxo modules, but get as close as possible.) - the gain settings for the fxs phone modules are intended to compensate for the loss associated with the cabling going to the attached phone plus any losses associated with the phone itself. Adjusting the fxs module gains to compensate for audio level problems associated with fxo modules will very likely cause other level problems (as you've probably already observed). The two module types (fxs and fxo) are totally indpendent and have nothing to do with the fact that both modules happen to be on the same TDM card. There is no logical reason that I can think of where an fxs module would need to be set at anything different then about 0 db. If you don't know what the analog pstn cable loss is between your asterisk box and the telco, then find that out _first_, and set the fxo gains to about 2 db less then that loss. (Example: if that loss is -8db, then set the rxgain txgain to about 6db.) The specific number will be a trade off between echo and being able to hear normal voice.) Your telco technical folks _might_ be able to tell you what that loss happens to be if you don't have the skills or tools to determine it for yourself. Once the fxo module gains are set to reasonable values, then muck with the fxs gains to help improve volume. If you don't follow the above, you will end up with one or more of the following: - calls from a sip phone to an fxs phone (with high gain settings) will be too loud and/or distorted - calls from the pstn to your asterisk system that roll to voicemail will be difficult (or impossible) to hear from any phone - calls to/from any itsp from your fxs phone will be too loud and/or distorted Think through the above before mucking with those gains. If the above doesn't make a lot of sense, search through the hundreds of postings that already exist on the topic and read more via the wiki. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mobilephone users get echo of them self when calling in to our asterisk server.
Hi there. The title basicly explains it. When we get incomming calls from cellular phones, the callers tend to echo ALOT. They hear their own voice at very high volums. This is a problem only for mobilphone users that calls in to us. Im using wifi IP-phones. Asteirsk: CVS-Nv1-0-7-04/19/05 Any way to fix this on the asterisk server? Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 301 second line registration,
Is your Asterisk server listening on port 5061? If not, just change the entry to 5060. On Wed, 2005-08-31 at 16:36 -0600, Andres Paglayan wrote: Hi, I am having problems on getting the second line to work on a Polycom 301, this is the phone.cfg file, the * box is 192.168.1.8 and the phone is 192.168.1.18 I am not 100% sure about what the reg.x.address should be, with this setting I only get the line number to work, the second just gives me busy signal, and its extension is not available. I also tried [EMAIL PROTECTED] and 203 as the reg.2.address parameter but without success, the 203 extension setting in Asterisk is a clon of the 200 except for the id and the port, (that matches this conf file) PHONE_CONFIG OVERRIDES reg.1.displayName=FD1 reg.1.label=L1 reg.1.address=192.168.1.18 reg.1.server.1.address=192.168.1.8 reg.1.server.1.port=5060 reg.1.auth.userId=200 reg.1.auth.password=123 reg.2.displayName=FD2 reg.2.label=L2 reg.2.address=[EMAIL PROTECTED] reg.2.auth.userId=203 reg.2.auth.password=123 reg.2.server.1.address=192.168.1.8 reg.2.server.1.port=5061/ /PHONE_CONFIG Thanks for any help, Andres ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
IS there a way to make the phone reboot each day at a time? On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) First, ensure that the 360 has Filter Packets from Registrar turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do sip show subscriptions from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP-2000 Poor sound Quality
I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again the static goes away, so I believe it is phone related. 2) When I call a non-voip phone when I stop talking (ie at the end of every sentence) the person on the other end hears some feedback/buzzing for a moment. Is anyone else using this phone and experiencing these issues? Has anyone else tried the GXP-2000 and decided to buy a different VoIP that they were impressed with (without spending too much money)? We have one GXP-2000 in house, and are trying to decided what phone to standardize on before we start rolling out them out to the users. Thanks, Aaron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What this little red cross mean in AAH
+ sign to add another trunk to the route Zeeshan Zakaria wrote: Hi, In asterisk at home, in Outbound Routing menu, under the trunk sequence (e.g. IAX2/FWD), what does little red cross mean beside the selected trunk. Thanks Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /etc/init.d/asterisk barfing
Sorry, went of at a tangent (that's what you get for half reading emails I guess!) Ok, guess the easiest thing to do is to check in the contrib directory, diff your one against the redhat (are you running redhat?) Mine is the same and it works fine, maybe you're running an outdated init script. Still worth trying the strace against the service command, at least it'll give you an idea of what it's trying to do. One last thing, did it work and then stop working or is this a fresh install and it's never worked? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 01 September 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P problems
On Thu, 2005-09-01 at 08:12 -0400, Andrew Kohlsmith wrote: On Thursday 01 September 2005 06:35, Alex Ongena wrote: I encounter 2 major problems: 1) Transmitting and receiving van Fax is very unreliable (on the CLI I see a Native bridging (seems to be 911.ulaw, 64 kbit high quality). Sometimes the Fax is Ok, sometimes I miss a few lines, sometimes it's impossible. The same problem is there when sending or receiving faxes. This is a well-known issue with the current drivers, although nobody's really stepped up to identify when exactly this started happenning (as the driver code was changed) or why. Do I understand that this would not be a problem with an 'older' version of Asterisk ? If so, any idea to which version I need to revert ? 2) I'm trying 3 different Analog phones and having 3 different behaviors: 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri ring... ri...) 1 phone does not ring at all when Asterisk says 'Ringing Zap/6'. However, when I do an 'off-hook' on this phone, I get a normal tone signal and I can dial and talk perfectly. DTMF is recognised too. It simply does not ring on incoming calls. Have you tried the 'boostringer=1' module option? If you swap phones and ports around (i.e. try phone #3 in phone #1's port) does the problem stay with the phone or the port? I was not aware of this option, I'll try it. The problem stays with the phone, regardless of the port? Thanks already alex -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
Yes. You could send a sip notify via asterisk and set that up via cron to happen once per day. eg. asterisk -rx sip notify reboot-snom sip peername the snom is at Make sure that the reboot-snom clause is setup in sip_notify.conf before attempting this. On 9/1/05, altus [EMAIL PROTECTED] wrote: IS there a way to make the phone reboot each day at a time?On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) First, ensure that the 360 has Filter Packets from Registrar turned off (under Advanced).Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor.With both of these in place, you should see an entry for each extension you are monitoring when you do sip show subscriptions from the * CLI.If not, rinse and repeat the above steps.Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users--Thanks Altus SnymanStormcorp Network Solutions+27 11 8071141 exten 301___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
If you do not want to buy a timer based power supply ;) you can send a NOTIFY with 'Event: reboot' to the phone. Nils On Thursday 01 September 2005 14:41, altus wrote: IS there a way to make the phone reboot each day at a time? On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote: PH I am setting up a snom 360, and the lights come on OK when the mapped PH user makes an outgoing call, but when the user takes an incoming call PH the light does not come on. PH I do not want to install the bristuff patch if possible. PH (although I can see that with the devstate command I can make the lights PH do whatever I want) First, ensure that the 360 has Filter Packets from Registrar turned off (under Advanced). Next, make sure you have hint priorities setup for each of the extensions you are trying to monitor. With both of these in place, you should see an entry for each extension you are monitoring when you do sip show subscriptions from the * CLI. If not, rinse and repeat the above steps. Also, you may want to manually recycle power on the 360 if you happen to reset asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to wait until the phones resend the subscription. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- snom technology AGGradestr. 46D-12347 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Strange problem with Bristuff
I tried to modify the extension but * soon receive the extension and the other hand get busy i.e First: The caller digit 123456 (at the fourth digit connect get the ring back) exten = _1234,1,dial,sip/25 exten = _123456,dial.sip/30 in this way the other hand get the first estension and does not pass the call to the extension After Caller digit 123456 (at the fourth digit get busy) (I deleted the first extension) exten = _123456,dial.sip/30 Any suggestion ? Stefan Tichy [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 01/09/2005 13.57 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 [Asterisk-Users] Re: Strange problem with Bristuff On Tue, Aug 30, 2005 at 07:45:50PM +0200, [EMAIL PROTECTED] wrote: I have a strange problem with a quadbri card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed. I have connected to the card 3 isdn in ptp mode configured in selection passing (I don't know if is exact the english traduction but I have 3 isdn with 99 numbers and asterisk forward the extensions) The problem is this: if I call from a cellular to asterisk all is Ok but when I try to call from a fixed line the extension (the last part of the number) is not sent but only the first part. It might be a pattern matching problem in your dial plan. The digits used as extension (0,..,9,00,...,99) might be transmitted one by one or as block. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommendation for 8 lines analog card in Australia
Where in Australia? We are based in Sydney. One of our clients have 4 analog lines using 2 dual FXO Cards and seems to work fine. Most of our clients have an E1. I Have not tried the 4 port cards however I would suggest 2 of them would be sufficient. OCTTEL have a 4 port router which is A-Ticked. They have an 8 port which would be perfect for you installation but not sure if that model have been A-Ticked yet: http://www.octtel.com.au/8440.htm Darren. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Thursday, 1 September 2005 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Recommendation for 8 lines analog card in Australia Hi, we want to build a Asterisk server for a branch office in Australia. At the moment they use 5 analog lines. We will need at least 8 lines. What hardware would you recommend for the 8 analog PSTN lines? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 or h323
I have just signed up for 2 landline numbers in China. They have offered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box. They have given me the following info (modified for security).. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Really what I want is for * to act as the endpoint. So the big question, do I use oh323 or h323 or something else. I am all confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all the incoming calls. Which one do I use and how would I tackle the conf file to register with the gatekeeper. Any help would be appreciated. Steve Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Micronet 5050s FXO gateway and hookflash transfers.
Hi, Has anyone out there managed to do a hookflash transfer with a Micronet 5050s gateway ? We have just tried out this gateway and it seems to do everything we need except this particular feature. Also if you have succeeded where is the hookflash time specified in the gateway. There does not appear to be any parameter for this. Perhaps it is not supported at all. Any help appreciated. regards, John. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues.conf OPTIONALURL within the Queues cmd
Hi Jason, Is there a specific version of DIAX that I should use? I grabbed the latest release...Looking at the DIAX site, 910g has the URL feature fixed. Is it broken again in 915a? URL feature works in 0.9.15a. Take care that it is implemented JUST for the Dial command. Best regards, Dan P.S. By the way. The latest version is 0.9.15b (with ATCOM phone support fixed) http://www.laser.com/dante/diax/diax0915b.zip Jason Walker wrote: I am testing this on * ver. 1.0.7 (I have another box with 1.0.9 and another one with CVS HEAD). Is 1.0.7 too old? Is this command not applicable to ver 1.0.7. That's probably your problem there. I know most newer versions of DIAX will do this. There is one of the later versions where the feature is broken. You probably need to update Asterisk. Kevin ___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 and hints
IS there a way to make the phone reboot each day at a time? You could do it via a cron job by wget'ting the reboot uri (on the advanced page again), but there really shouldn't be any need to do so. The only time subscriptions should disappear is when you do a reload or restart on asterisk. Even after a reload or restart the subscriptions will come back, but it usually takes ~30 minutes or so depending on when the last subscriptions were sent. -Jeff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Asterisk to a Toshiba Strata system
I have a Digital Toshiba Strata PBX with an IP Card (Model BIPU-M2A) that supports 16 connections and I want to Integrate Asterisk via this card but I found out it supports MEGACO+(I believe share a lot with MGCP). I am not sure how to go forward with this. (Note: Theproprietary IP Phones work via this card) I am familiar with with IAX, SIP but no MGCP but If anyone has any idea I am willing to study it. Can anyone help me? I also need clarification a statement I read somewhere that Asterisk is Call Agent not and a User Agent when it comes to MGCP Thanks, Michael A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] /etc/init.d/asterisk barfing
Not sure this applies, but there does seem to be a problem in the zaptel area that also impacts service... command. If I kill asterisk and try 'service zaptel stop', it now fails consistently as its unable to stop 'ztdynamic' (leaving zaptel still running since it can't unload). However, 'service zaptel start' does succeed on my FC3 cvs-head (from Aug 25th) and one TDM card installed. Sorry, went of at a tangent (that's what you get for half reading emails I guess!) Ok, guess the easiest thing to do is to check in the contrib directory, diff your one against the redhat (are you running redhat?) Mine is the same and it works fine, maybe you're running an outdated init script. Still worth trying the strace against the service command, at least it'll give you an idea of what it's trying to do. One last thing, did it work and then stop working or is this a fresh install and it's never worked? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 01 September 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See
[Asterisk-Users] Re: Polycom 301 second line registration
Is your Asterisk server listening on port 5061? If not, just change the entry to 5060. Also, I'm not sure how your sip.conf is set up for asterisk, but if you've set it up like: [203] type=friend username=blah secret=blah etc... Your Polycom config file will generally look like this. PHONE_CONFIG OVERRIDES reg.1.displayName=FD1 reg.1.label=L1 reg.1.address=203 reg.1.server.1.address=192.168.1.8 reg.1.server.1.port=5060 reg.1.auth.userId=blah reg.1.auth.password=blah reg.2.displayName=FD2 reg.2.label=L2 reg.2.address=203 reg.2.auth.userId=blah reg.2.auth.password=blah reg.2.server.1.address=192.168.1.8 reg.2.server.1.port=5060/ /PHONE_CONFIG - Noah On Wed, 2005-08-31 at 16:36 -0600, Andres Paglayan wrote: Hi, I am having problems on getting the second line to work on a Polycom 301, this is the phone.cfg file, the * box is 192.168.1.8 and the phone is 192.168.1.18 I am not 100% sure about what the reg.x.address should be, with this setting I only get the line number to work, the second just gives me busy signal, and its extension is not available. I also tried [EMAIL PROTECTED] and 203 as the reg.2.address parameter but without success, the 203 extension setting in Asterisk is a clon of the 200 except for the id and the port, (that matches this conf file) PHONE_CONFIG OVERRIDES reg.1.displayName=FD1 reg.1.label=L1 reg.1.address=192.168.1.18 reg.1.server.1.address=192.168.1.8 reg.1.server.1.port=5060 reg.1.auth.userId=200 reg.1.auth.password=123 reg.2.displayName=FD2 reg.2.label=L2 reg.2.address=[EMAIL PROTECTED] reg.2.auth.userId=203 reg.2.auth.password=123 reg.2.server.1.address=192.168.1.8 reg.2.server.1.port=5061/ /PHONE_CONFIG Thanks for any help, Andres ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: equipment configuration help
Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have to buy an analog card? or maybe link me to a web site that explains (with images) how a t1/e1 is managed? Thanks, and I apologize for this completely newbie question. I've never seen images or instructions on how to handle this. Im not even sure im using the right terms in Google. -- --- Erick Perez Linux User 376588
Re: [Asterisk-Users] Re: equipment configuration help
Erick- Can't say if they will or not. In theory they should respect all outgoing traffic unless being filtered by another device such as your PIX. You might want to check with the ADSL router manufacturer just to be safe. On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote: Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have to
[Asterisk-Users] Overhead Paging Systems...
Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything like this? This is my first * installation, and I havn't been finding too much on google that helps me. ~kurth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 hold problem
Hello, I have a customer who said that their Snom 360 is joining calls by accident. The situation is that they had one call on the line and another call came in. She pressed the hold button on the phone and the two calls were joined together. I do have Call join on Xfer set to yes, but I thought that would only come into play when doing a transfer, not putting someone on hold. The phone is at firmware 4.1, and there are no new updates, so that shouldn't be it. Anyone else experience this behavior on the phones, or know if I need to turn off Call Join on Xfer? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP - Queue Transfer
I have two call queues in which the agents are added through AgentCallbackLogin. Whenever they answer the call then TRANSFER, the call is transferred but the agent status continues to show them talking on the zap channel. Eventually, timeout with no RTP traffic the agent appears to be transferred to the new extesion. Once the call is terminated by the person to whom it is transferred the agent returns to their appropriate context and extension as shown in agent status. The same result is had for sip or PBX transfers. I am running asterisk stable. thanks, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for Russia and Israel Dids
hello I am looking for Israel and Russia DiDs. Please email me to [EMAIL PROTECTED] REgards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 or h323
Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box.They have given me the following info (modified for security)..Protocol = H323Gatekeeper = 210.21.118.xxxH323ID = .HMA0200.10szxn-hxxxe164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxxe164 = 02022xx2913Really what I want is for * to act as the endpoint.So the big question, do I use oh323 or h323 or something else. I amall confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all theincoming calls.Which one do I use and how would I tackle the conf file to registerwith the gatekeeper.Any help would be appreciated. Steve Ducat.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zapata nationalprefix-problem [Virus checked]
has anyone an idea how to display incoming national/international isdn-pstn-calls correctly to internal isdn AND sccp/sip-phones ? without nationalprefix=0 and internationalprefix=00 I get incoming phone numbers correctly on isdn-phones but the leading zero's are stripped of for non-isdn phones when I set this prefixes inside zapata.conf my internal isdn-phones get this prefix twice... is it possible to unset the prefixes for one or more cards serving internal line ? I tried it without luck ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /etc/init.d/asterisk barfing
service zaptel start|stop works fine for me. service asterisk start does not [EMAIL PROTECTED] res]# service asterisk start Starting asterisk: [ OK ] [EMAIL PROTECTED] res]# service asterisk status asterisk dead but pid file exists [EMAIL PROTECTED] res]# This is on a up-to-date CentOS 4 box. The service command *was* working before I tried to install 1.2beta :( Julian. Rich Adamson wrote: Not sure this applies, but there does seem to be a problem in the zaptel area that also impacts service... command. If I kill asterisk and try 'service zaptel stop', it now fails consistently as its unable to stop 'ztdynamic' (leaving zaptel still running since it can't unload). However, 'service zaptel start' does succeed on my FC3 cvs-head (from Aug 25th) and one TDM card installed. Sorry, went of at a tangent (that's what you get for half reading emails I guess!) Ok, guess the easiest thing to do is to check in the contrib directory, diff your one against the redhat (are you running redhat?) Mine is the same and it works fine, maybe you're running an outdated init script. Still worth trying the strace against the service command, at least it'll give you an idea of what it's trying to do. One last thing, did it work and then stop working or is this a fresh install and it's never worked? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 01 September 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in
Re: [Asterisk-Users] Oracle Realtime Driver and CDR Logger
Chris Deserva wrote: I have written it in C++, because I used an OCI interface library (ORAPP). I want to post it opensource so that I could get help in its development and testing, and be a part of Asterisk modules. You cannot make this open source. The Oracle client libraries are not license-compatible with open source licenses, so it's not legal for you to distribute code which links to them and is open source. Obviously that also means it cannot be part of Asterisk as a standard module. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE cards with ISDN BRI?
Aaron Picht wrote: Does anybody know if the Digium TE series cards will work with NI-1 (SBC California) ISDN BRI? If not can anyone make recommendations as to reliable cards to use? My end goal is to use the BRI lines for incoming fax (spandsp) only. No Digium boards work with BRI circuits directly. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging Systems...
The two most common companies to make paging equipment are Viking and Bogen. Bogen even resells ATAs for paging now. http://www.bogen.com or http://www.vikingelectronics.com Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, September 01, 2005 7:33 AM Subject: [Asterisk-Users] Overhead Paging Systems... Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything like this? This is my first * installation, and I havn't been finding too much on google that helps me. ~kurth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323 or h323
I'm using oh323 too without any issues, but in Steve specific configuration, depends on how his provider expect to be register as (Terminal or Gw) afaik, oh323 just could be binded as gateway, so better ask the provider. LTenorio From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mehdi chouikhSent: Thursday, September 01, 2005 12:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] oh323 or h323 Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box.They have given me the following info (modified for security)..Protocol = H323Gatekeeper = 210.21.118.xxxH323ID = .HMA0200.10szxn-hxxxe164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxxe164 = 02022xx2913Really what I want is for * to act as the endpoint.So the big question, do I use oh323 or h323 or something else. I amall confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all theincoming calls.Which one do I use and how would I tackle the conf file to registerwith the gatekeeper.Any help would be appreciated. Steve Ducat.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk with Meridian1 option11 in the UK
Hi, The PBX recives alarms from the TE110p card and are mainly pointing at frame errors and Loss of signal. Asterisk is configured as Zapata.conf signalling=pri_cpe switchtype=national rxwink=250 channel = 1-15,17-31 Zaptel.conf This is what I need to know - the SPAN is currently set to E3 - does anyone know what I need to use for a E1 ? span=1,1,0,esf,b8zs bchan=1-15,17-31 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 loadzone= uk defaultzone = uk I am currently trying to connect a Meridian1 OP11 (card in the meridian box is a NTBK50AA 2mbPRI (E1)) with a asterisk server (TE110P card) however we are experiencing problems in the communication between the two. The pin outs are as follows PRI CARD PINS (rj45) 1 w/orange 2 orange 3 4 blue 5 w/blue 6 7 8 NTBK50AA has a Telco 50 connector PROVIDER PINS - as per the op11 DTI/PRI install/maint manual 48 blue 24 w/orange 23 orange 49 w/blue Any help will be greatly appreciated Thanks Chandra Mistry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enable anonymous SIP incoming calls
Hi all, I would like to enable SIP incoming calls from any origin (not configured as peer in sip.conf). Is this possible? A workaround could be to put a SER in front of the Asterisk, and configure this as a peer in sip.conf, but I would like to find other simpler way if possible. Thank you very much. G. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astaro SIP Proxy
Hi, Did anyone successfully installed and setup Astaro Security Linux V6 SIP Proxy with Asterisk behind the Astaro and clients bedinde another NAT? Regards Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] detecting extensions in use
Michiel van Baak wrote:
On 17:22, Wed 31 Aug 05, Eric Skippy Hope wrote:
We're using 1.0.9 and the powers that be are wary of moving beyond
stable. If I'm reading the wiki correctly, incominglimit is to limit
the calls coming _from_ the extension and coming into the server, and
outgoinglimit is commented out in the source code. The recomendation is
to use SetGroup and CheckGroup for this, but they don't work correctly
when ringing multiple lines.
I'd be happy to loop through all of the possible extensions, check each
one to see if it has a call, and if not put into a variable to be dialed
at the end, but how do I tell if an extension is involved in a call?
Hi,
Did you read that page totally?
I guess not! :)
I think the trick with the local/ construct with /n at the
end can be the solution.
We use this to check every extension against our Groupware's
calendar database to see if a user has a meeting and doesn't
want to take calls.
Dial(Local/[EMAIL PROTECTED]/nLocal/[EMAIL PROTECTED]/n);
That way we can do totally different stuff depending on the
extension in the contect internalphones.
You should be able to do a setgroup/checkgroup on all the
local channels :)
Thats exactly what I needed. Thank you very much. You've saved me from
a headache.
If anyone else can use it, the section looks like:
[macro-allextens];
exten = s,1,SetGroup(${ARG1}ACTIVE)
exten = s,2,CheckGroup(1)
exten = s,3,Dial(${ARG1},120)
exten = s,4,Hangup
exten = s,104,Playtones(busy)
exten = s,105,Busy
[sales-line]
exten = s,1,SetCIDName(SALES)
exten =
s,2,Dial(Local/[EMAIL PROTECTED]/nLocal/[EMAIL PROTECTED]/nLocal/[EMAIL PROTECTED]/nLocal/[EMAIL PROTECTED]/nLocal/[EMAIL PROTECTED]/n)
exten = x01,1,Macro(stdexten-test,${X301})
exten = x02,1,Macro(stdexten-test,${X302})
exten = x03,1,Macro(stdexten-test,${X303})
exten = x04,1,Macro(stdexten-test,${X304})
exten = x05,1,Macro(stdexten-test,${X305})
-skippy
___
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[Asterisk-Users] Loop error when compiling CVS version of 1.2-Beta
I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c ast_expr2.c ast_expr2f.c asterisk.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c This is the contents of the include/asterisk/version.h file. /* * version.h * Automatically generated */ #define ASTERISK_VERSION CVS-Nv1-2-0-beta1-09/01/05-11:06:01 #define ASTERISK_VERSION_NUM 99 Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Authentication
I want to be able to authenticate each user that dial out (PSTN or IP), ideally using their mailbox and voicemail password. Should I use VMAuthenticate or Billing? Any pointers on setting up either? Cheers Graham___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality
Greetings We have all those problems and then some... after a while, the phone starts degrading: The ringing becomes lower and lower and there is a lot of stuttering in the conversation. Also, if I stop/start asterisk, half of the phones reconnect while the rest don't. I was using the same firmare as you but had to roll back to 1.0.1.9 because of the degrading issue. We have some polycoms connecting to the same server and they have no problems whatsoever so we know it's a problem with the GXP. These phones are definately NOT ready for prime time. I would stay away from them. Play it safe and use Polycoms or, if too expensive, maybe Sipuras 841. These GXP-2000s are pure evil. Jesus Mogollon Global IP Systems, LLC http://www.globalipsystems.com2005/9/1, Aaron W [EMAIL PROTECTED]: I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again the static goes away, so I believe it is phone related. 2) When I call a non-voip phone when I stop talking (ie at the end of every sentence) the person on the other end hears some feedback/buzzing for a moment. Is anyone else using this phone and experiencing these issues? Has anyone else tried the GXP-2000 and decided to buy a different VoIP that they were impressed with (without spending too much money)? We have one GXP-2000 in house, and are trying to decided what phone to standardize on before we start rolling out them out to the users. Thanks, Aaron ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Overhead Paging Systems...
Viking makes everything you might need for paging and door control. www.vikingtelecomsolutions.com William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, September 01, 2005 7:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Overhead Paging Systems... Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything like this? This is my first * installation, and I havn't been finding too much on google that helps me. ~kurth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/86 - Release Date: 8/31/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/86 - Release Date: 8/31/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One way echo canceling?
On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote: When there is a call on zap 1, from a sip phone on the remote office side and typing 'zap show channel 1' shows echo cancel is on, doing the same thing from the Definity to a SIP phone shows echo cancel off. Shouldn't it be on during a call on both the incoming and outgoing legs as long as it comes accross the PRI? Some (Myself included) have noted a slight echo on the Definity to SIP leg of the connection. My zapata.conf is below: switchtype = national context = incoming signalling = pri_cpe echocancel=yes echotraining = yes echocancelwhenbridged=yes overlapdial = yes group = 1 channel = 1-23 I have not seen it myself, but I have heard that some people have ahd trouble with overlapdial and echo cancellation. I have not been able to confirm whether or not this is actually a bug. One possible fix is to disable overlapdial and see if echo cancellation is enabled after this. If it is, this might be a bug in chan_zap.c -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oracle Realtime Driver and CDR Logger
And to add to what Kevin said, we don't want any closed source stuff, be it a database module or a device driver, to be a part of Asterisk as a standard module, for obvious reasons. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, September 01, 2005 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Oracle Realtime Driver and CDR Logger Chris Deserva wrote: I have written it in C++, because I used an OCI interface library (ORAPP). I want to post it opensource so that I could get help in its development and testing, and be a part of Asterisk modules. You cannot make this open source. The Oracle client libraries are not license-compatible with open source licenses, so it's not legal for you to distribute code which links to them and is open source. Obviously that also means it cannot be part of Asterisk as a standard module. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One way echo canceling?
Matt Fredrickson wrote: On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote: When there is a call on zap 1, from a sip phone on the remote office I have not seen it myself, but I have heard that some people have ahd trouble with overlapdial and echo cancellation. I have not been able to confirm whether or not this is actually a bug. One possible fix is to disable overlapdial and see if echo cancellation is enabled after this. If it is, this might be a bug in chan_zap.c Turning off overlapdial did indeed fix it. It now shows as being enabled. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM04b and echo
On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote: the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am currently considering (and would like to know if these might be useful) 1 upgrade from 1.09 ( asterisk at home ) to 1.2 cvs code base That is worth a shot. There are a few new echo-related features that have been added: 1.) fxotune - try this first. There is a file called README.fxotune that explains how to use it. It is primarily for doing echo related line tuning (which in your case possibly won't help). 2.) Also, there is a new echo canceller in CVS-HEAD zaptel that has received a lot of positive feedback. Look in zconfig.h for ECHO_CAN_KB1 for further information. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztcfg problem
Hi all, Im installing two HFC pci card (both in TE mode), I dont have problem when load module, but whrn I give ztcfg vv, I see 6 the six channels that I configured, then my computer hang and I have to reboot it. (Im using a VIA Epia-M 1000 with Via C3 processor) [EMAIL PROTECTED]:~# modprobe zaptel [EMAIL PROTECTED]:~# insmod /usr/src/bristuff-0.2.0-RC8n/zaphfc/./zaphfc.o [EMAIL PROTECTED]:~# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) 6 channels configured. Any ideas ? Thanks all Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing extension, which context is searched
Hi When a a context , and you dial a extension, will the current one be looked into, or the default incoming one. My call scenario is to bring in all users into a default context and then GoTo others based upon some parameter. Now when a user dials a extension, the match should occur within the extension he went to in the GoTo, or will it match in the context he was sent from. Iqbal ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma card problem with EWSD Exchange
On Thu, Sep 01, 2005 at 01:24:02AM -0500, Kristian Kielhofner wrote: Nguyen Trung Tin wrote: I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. Have you tried contacting Sangoma? I forwarded this to them. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *66 with Sipura devices.
Has anyone gotten *66 (busy callback) to work with asterisk and devices like the sipura SPA-2002? When I dial *66 and hangup.. the sipura seems to immediately try again (which is probably normal) and then ring me (Even if the line is busy) any idea why the sipura is not detecting the line as busy? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overhead Paging Systems...[More Info]
Thanks to all that have replied thus-far. I have talked to Viking and they recommended the PA-2A paging amp. (http://www.vikingelectronics.com/products/view_product.php?pid=317). It requires a 600ohm input. Before I go beating my head against the wall with this, has anyone else installed something similar? Can you tell me what hardware you used? Any help would be better then none. Thank You! ~kurth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to resolve SMS/WAP/MMS/VoIP gateways on a shoestring?
I was wondering if anyone could shed some light on what options I have for mapping incoming/outgoing SMS messages to/from a telephone number that I am given by a VoIP provider who does not currently offer SMSC services? In other words, Voicepulse, my VoIP provider, provides me with a PSTN terminated number (hypothetically 222-222-). I use my Asterisk server to handle the calls that Voicepulse delivers to me via IAX. I need to send/receive SMS messages to/from the mobile carrier network. As far as I understand it I need a SMSC provider with access to the mobile carrier network. I currently have Asterisk triggering Kannel to send SMS messages. I am on a shoestring budget so registering for a ShortCode number is out of the question. My objective is to handle incoming voice calls at 222-222- *AND* be able to send/receive SMS messages from 222-222- without playing a game of call forwarding or faking my SMSC ID. Right now, my mobile telephone (hypothetically 777-777-) is my SMSC. I have opted to go this route because of the cheap, unlimited SMS/data plans that are available. Ultimately, I want to be able to send/receive/respond to SMS messages at the same number that I route my VoIP service to -- currently, if you call 222-222- you get a SMS from 777-777-. Obviously, this is not ideal. Down the road, I will want to be able to have my MMS gateway, Mbuni, process incoming MMS messages sent to my VoIP telephone number. This is becoming an issue that I'd like to understand better. I'd also like to solve bridging my routing issues, in an affordable manner (or at least understand what's possible even if I can't afford it.) It would be nice if there was a way to setup something like a CNAME record which would allow me to resolve my various gateways. I suspect that having a 5-digit .sc is the solution to all my problems but I am a bit unclear how that stuff works. Can anyone shed light on what options I might have to resolve the issues I bring up? In particular, how can I setup my SMS/MMS/WAP/VoIP gateways so that they map each of their respective services, and resolve everything that is directed to them, to/from the same number? I'm having a hard time explaining this complicated situation which I have only an intermediate understanding of in the first place. Thank you for helping me get a better sense of they way these things work, and hopefully can work better, together. As I develop my VoIP services I need affordable solutions to allow me to continue my rd work. This is why cost is such a major issue and SC is out. Thank you for your help clarifying things for me. Cheers, HJ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E-Tech ADWV01
Does anybody have experience with this all-in-one router? (triple play, ADSL, VoIP, Wifi). I needed to upgrade the firmware, or otherwise the ADSL Internet connection would drop during a call. This has been fixed now. Also the device registers fine with my asterisk. Incoming calls through the FXO port work. When an incoming call via SIP comes in, the phone rings and then the SIP connection is dropped with status: NO-ANSWER. How do I make it work? My other question: Does somebody have a working example so a Caller ID signal is presented on the FXS ports? TIA, Rene Kluwen Chimit ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with include
Hi, I put on sip.conf the following line #include sip.d/*.conf inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this is the ls result [EMAIL PROTECTED] asterisk]# ls /etc/asterisk/sip.d/ -la total 13 drwxrwxrwx 2 asterisk asterisk 4096 Sep 1 13:06 ./ drwxr-xr-x 9 asterisk asterisk 4096 Sep 1 13:17 ../ -rwxrwxrwx 1 asterisk asterisk 276 Sep 1 13:06 provider1.conf* -rwxrwxrwx 1 asterisk asterisk 274 Sep 1 13:06 provider2.conf* ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Buying DIDs
Other than DIDx what is another DID provider that I can buy DIDs from? Joshua ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED]: How to changed AMP User Login andPassword
Hi, you can get the following by typing 'help-aah' from the CLI Commands Descriptions---config set the local time zone and keyboard typenetconfig configure ethernet interfacegenzaptelconf autoconfig Zaptel cardsrestore-aah restore from a backupinstall-AVMB1ISDN install support for AVB B1 ISDN cardinstall-EiconDiva install support for Eicon Diva ISDN cardinstall-pdf installs support for emailing PDFs of faxespasswd-maint set master password for web GUIpasswd-amp set password for amp onlypasswd-meetme set password for Web MeetMe onlypasswd set root password for console loginpasswd admin set admin password for checking system mailsetup-cisco create a SIPDefault.cnf in /tftpbootsetup-dhcp set up a dhcp serverrebuild_zaptel rebuild zaptel driver after kernel updateasterisk -r Asterisk CLIyum -y update Get latest patches for CentOS thanks Chands From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan ZakariaSent: 01 September 2005 03:36To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] [EMAIL PROTECTED]: How to changed AMP User Login andPassword Hi, I cant figure out how to change User Login and Password for AMP. By default it is user:admin and password:maint. Anybody knows how to do it? Zeeshan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality
Thanks..I am begining to agree with you about these phones. Which poylcoms do you have? I have been looking at the polycom soundpoint IP501. It seems like a good phone for just under 200USD. Thanks again, Aaron On 9/1/05, Jesus Mogollon [EMAIL PROTECTED] wrote: Greetings We have all those problems and then some... after a while, the phone starts degrading: The ringing becomes lower and lower and there is a lot of stuttering in the conversation. Also, if I stop/start asterisk, half of the phones reconnect while the rest don't. I was using the same firmare as you but had to roll back to 1.0.1.9 because of the degrading issue. We have some polycoms connecting to the same server and they have no problems whatsoever so we know it's a problem with the GXP. These phones are definately NOT ready for prime time. I would stay away from them. Play it safe and use Polycoms or, if too expensive, maybe Sipuras 841. These GXP-2000s are pure evil. Jesus Mogollon Global IP Systems, LLC http://www.globalipsystems.com2005/9/1, Aaron W [EMAIL PROTECTED]: I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again the static goes away, so I believe it is phone related. 2) When I call a non-voip phone when I stop talking (ie at the end of every sentence) the person on the other end hears some feedback/buzzing for a moment. Is anyone else using this phone and experiencing these issues? Has anyone else tried the GXP-2000 and decided to buy a different VoIP that they were impressed with (without spending too much money)? We have one GXP-2000 in house, and are trying to decided what phone to standardize on before we start rolling out them out to the users. Thanks, Aaron ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TOS bit and DSCP
Good Day all, I have a box connected to a netgear switch which allows me to set priority based upon DSCP Values. This switch has listings from value 1 - 63. And can be set to normal, high, etc. Does anyone know what or how to translate TOS= line in the sip.conf file to in order to have this switch prioritize voip data? Thanks in advance ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk run problem, was working, rebooted server, now nothing
great nevermind. I was relying on my friend's configurations which are broken at agent-add. He was logging in the phones by just going through typing in each login manually on each phone. :| I'm curious then why the heck he placed a login key on each phone. ... :( Original Message: - From: [EMAIL PROTECTED] Date: Thu, 1 Sep 2005 3:33:52 -0400 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk run problem, was working, rebooted server, now nothing I've copied over all the asterisk configuration settings. There was nothing decent to see in the logs, so I#12288;didn't copy those. http://zanshin.tsumelabs.com/ The system was working a couple days ago until I had the server rebooted. there are 2 zap cards, both are working fine, all lines are able to recieve phone calls. The problem is callers call the building and they are intruduced to the weekday message, then when they hit 1 they are placed in a queue for appointments. They are always on hold and the receptionist phone doesn't ring. What could be the problem? receptionist == 101 Tsume mail2web - Check your email from the web at http://mail2web.com/ . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip jitter buffer in 1.2?
Did the sip jitter buffer make it into 1.2? anyone using it? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop error when compiling CVS version of 1.2-Beta
sorry to report that I reported exactly the same error, twice. Never got a response from anybody on the 1.2beta team. CVS head compiles just fine. Julian. Geoff Karl wrote: I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c ast_expr2.c ast_expr2f.c asterisk.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c This is the contents of the include/asterisk/version.h file. /* * version.h * Automatically generated */ #define ASTERISK_VERSION CVS-Nv1-2-0-beta1-09/01/05-11:06:01 #define ASTERISK_VERSION_NUM 99 Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax trouble with HP 3330mfp (again)
I'm using Asterisk 1.09 with bristuff 0.2.0-RC8n, one BRI line and a
Sipura SPA-2000. I have a HP LaserJet 3330mfp all-in-one. I can receive
faxes but not send them. The faxes start whistling to each other but the
transmission is stopped with a communication error
To receive a fax I have this in my dialplan:
exten = 00,1,Ringing
exten = 00,2,zapEC(off) ; disable EC on the incoming channel
exten = 00,3,SetCIDNum(${PRI_NETWORK_CID})
exten = 00,4,LookupCIDName
exten = 00,5,Dial(${FAX})
To send a fax I use these options:
exten = _9.,1,Dial(ZAP/g1md/${EXTEN:1},70,rdT)
exten = _9.,2,Macro(fastbusy)
I'm puzzled why I can receive faxes but not send them. The HP is capable
of 14k4 transmission speeds (and I think even higher) but why wouldn't
that be a show stopper to receive faxes?
In the firmware of the HP I cannot discover an option to limit the tx/rx
speed.
Any hints on what I am doing wrong greatly appreciated. Is there another
way to force the faxes to try 9k6 as the max speed or does my dialplan
have mistakes?
Thanks!!
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Re: [Asterisk-Users] TDM04b and echo
Matt Fredrickson wrote: On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote: the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am currently considering (and would like to know if these might be useful) 1 upgrade from 1.09 ( asterisk at home ) to 1.2 cvs code base PS you dont' need to upgrade asterisk to CVS Head to use the Zaptel from CVS head and the new Echo setup. I just installed it on 3 systems and they all improved. Using the KB1 That is worth a shot. There are a few new echo-related features that have been added: 1.) fxotune - try this first. There is a file called README.fxotune that explains how to use it. It is primarily for doing echo related line tuning (which in your case possibly won't help). 2.) Also, there is a new echo canceller in CVS-HEAD zaptel that has received a lot of positive feedback. Look in zconfig.h for ECHO_CAN_KB1 for further information. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000 Poor sound Quality
On Thu, 1 Sep 2005, Jesus Mogollon wrote: We have all those problems and then some... after a while, the phone starts degrading: The ringing becomes lower and lower and there is a lot of stuttering in the conversation. Also, if I stop/start asterisk, half of the phones reconnect while the rest don't. I was using the same firmare as you but had to roll back to 1.0.1.9 http://1.0.1.9 because of the degrading issue. We have some polycoms connecting to the same server and they have no problems whatsoever so we know it's a problem with the GXP. These phones are definately NOT ready for prime time. I would stay away from them. Play it safe and use Polycoms or, if too expensive, maybe Sipuras 841. These GXP-2000s are pure evil. In fairness the 1.0.1.9 firmware works very well for us. The speakerphone has an unusable microphone, but that is not an issue for us. Other than that we have not experienced any problems. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with include
Il Neofita wrote: Hi, I put on sip.conf the following line #include sip.d/*.conf You neglected to include the most important piece of information: what version of Asterisk you are using. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speed Questiosn
Hi I currently have a 3072kbps line that I'm splitting in half for 5 of my phones. That's 307.2kbps +/- a couple of kpbs. What is the minimum kbps for a phone to maintain clarity and volume? Joshua ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialparties.agi is returning no extensions to dial
Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if they press 1 : exten = 1,1,Goto(ext-group,3,1); goto ringgroup 3, the sales group In [ext-group] I have these lines in [ext-group] to define the sales ringgroup: exten = 3,1,Setvar(GROUP=103) ; the Sales group is group 3 -- only Dick is in for now exten = 3,2,Setvar(RINGTIMER=30) ; rings for 30 seconds max exten = 3,3,Setvar(PRE=Sales) ; called id has Sales: pre-pended exten = 3,4,Macro(rg-group); rings the group exten = 3,5,Macro(vm,103,1); goes to Dick's voice mail if no-one picks up I believe that this should cause extension 103 to ring, and then if it isn't picked up it will go to 103's voicemail ( eventually I will add other extensions to the group, but leave the fall-through to go to 103's vm). What happens when I call in from the outside is that the call goes directly to the voicemail of 103. Here are some logs from the *CLI prompt and from the /var/log/asterisk/full file: from the *CLI Don't know what to do if second ROSE component is of type 0x6 -- Accepting call from '512xxx' to '5126xxx' on channel 0/1, span 1 -- Executing Goto(Zap/1-1, aa_default|s|1) in new stack -- Goto (aa_default,s,1) -- Executing GotoIf(Zap/1-1, 0?4) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, DIR-CONTEXT=ext-local) in new stack -- Executing DigitTimeout(Zap/1-1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout(Zap/1-1, 7) in new stack -- Set Response Timeout to 7 -- Executing BackGround(Zap/1-1, custom/aa_default) in new stack -- Playing 'custom/aa_default' (language 'en') == CDR updated on Zap/1-1 -- Executing Goto(Zap/1-1, ext-group|3|1) in new stack -- Goto (ext-group,3,1) -- Executing SetVar(Zap/1-1, GROUP=103) in new stack -- Executing SetVar(Zap/1-1, RINGTIMER=30) in new stack -- Executing SetVar(Zap/1-1, PRE=Sales) in new stack -- Executing Macro(Zap/1-1, rg-group) in new stack -- Executing GotoIf(Zap/1-1, 0?3:2) in new stack -- Goto (macro-rg-group,s,2) -- Executing SetCIDName(Zap/1-1, AIRLINK SYSTEMS) in new stack -- Executing SetVar(Zap/1-1, RGPREFIX=) in new stack -- Executing SetCIDName(Zap/1-1, AIRLINK SYSTEMS) in new stack -- Executing SetVar(Zap/1-1, RecordMethod=Group) in new stack -- Executing Macro(Zap/1-1, record-enable|3|Group) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(Zap/1-1, 0?5:8) in new stack -- Goto (macro-record-enable,s,8) -- Executing GotoIf(Zap/1-1, 1?9:12) in new stack -- Goto (macro-record-enable,s,9) -- Executing AGI(Zap/1-1, recordingcheck) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck: Extension List not set -- nothing to do -- AGI Script recordingcheck completed, returning 0 -- Executing SetVar(Zap/1-1, CALLFILENAME=g3-20050901-115459-1125593688.105) in new stack -- Executing Goto(Zap/1-1, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(Zap/1-1, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(Zap/1-1, NO RECORDING NEEDED) in new stack -- Executing Macro(Zap/1-1, dial||tr|) in new stack -- Executing GotoIf(Zap/1-1, 1?4:2) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = 5122311245 -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 33 -- dialparties.agi: dnid = 5126873305 -- dialparties.agi: request = dialparties.agi -- dialparties.agi: calleridname = AIRLINK SYSTEMS -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: uniqueid = 1125593688.105 -- dialparties.agi: callingpres = 3 -- dialparties.agi: type = Zap -- dialparties.agi: rdnis = unknown -- dialparties.agi: callingtns = 0 -- dialparties.agi: enhanced = 0.0 dialparties.agi: Caller ID name and number are '5122311245' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Not found (No such file or directory) == Manager 'admin' logged on from
[Asterisk-Users] Two devices behind nat
Hello Everyone, I have one machine (asterisk server) that is DMZ behind my nat firewall on my client end (at home) i have a linksys wrt54g with 16384-32766 forwarded to my cisco 7960 (which works fine) and 16000 - 16383 forwarded to my sipura 2100 (which is set to these ports) For some reason, the sipura only works some of the time. Both devices are set to register, is there anything else I need to do to get them to work behind nat? Thanks! Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low handset microphone volume with Sipura SPA-841
Just in case somebody else has this problem, it seems that there is a bug in the 3.1.3a version of firmware of the Sipura SPA-841. Updating to the 3.1.4a version of the firmware solved the problem. On Sun August 28 2005 01:55, Juan Jose Comellas wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TOS bit and DSCP
this is a link where you can understand the relationship between tos and dscp http://www.speedguide.net/tcpoptimizer.php Ros - Original Message - From: Ronald Hartmann [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, September 01, 2005 1:37 PM Subject: [Asterisk-Users] TOS bit and DSCP Good Day all, I have a box connected to a netgear switch which allows me to set priority based upon DSCP Values. This switch has listings from value 1 - 63. And can be set to normal, high, etc. Does anyone know what or how to translate TOS= line in the sip.conf file to in order to have this switch prioritize voip data? Thanks in advance ~ron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip jitter buffer in 1.2?
I am using it with CVS-HEAD but it is currently a patch. So far the version of the patch I have (which was the first one released).. seems to be working very well.. and definately makes a noticeable improvement. On 9/1/05, Damon Estep [EMAIL PROTECTED] wrote: Did the sip jitter buffer make it into 1.2? anyone using it? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to speed-up INCOMING-RINGING-ENDED detection on X101P/zapata?
Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time (4.5 sec) for incoming ring end detection? I measured, event -- Hungup 'Zap/1-1' is shown exactly 8 sec after last detected ring (on X101P), and my voip phone continues to ringing during that time (that's bad). I want to cut that time to 4.5 sec. How to do that? I tried to change in zapata.h some lines: #define ZT_DEFAULT_RINGTIME 500 #define ZT_LOOPCODE_TIME 3000 #define ZT_RINGOFFTIME 2000 but with no effects. Hungup is still shown 8 sec after last ring. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg problem
On Thu, Sep 01, 2005 at 06:47:10PM +0200, Giordano Grandis wrote: Hi all, I'm installing two HFC pci card (both in TE mode), I don't have problem when load module, but whrn I give ztcfg -vv, I see 6 the six channels that I configured, then my computer hang and I have to reboot it. (I'm using a VIA Epia-M 1000 with Via C3 processor) When exactly does it hang? What is your zapata.conf? Do you run it from the console? if so: do you see an oops trace? Is the system totally hung, or can you still get useful information from alt-sysrq-p and such? [EMAIL PROTECTED]:~# modprobe zaptel [EMAIL PROTECTED]:~# insmod /usr/src/bristuff-0.2.0-RC8n/zaphfc/./zaphfc.o And which version of zaptel? And linux? [EMAIL PROTECTED]:~# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) 6 channels configured. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /etc/init.d/asterisk barfing
On Thu, Sep 01, 2005 at 10:15:04AM +0100, Julian Lyndon-Smith wrote: Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) The command that is actually run at startup is something like: /etc/rc3.d/S30asterisk start And it is actually run from an environment that is slightly different from your standard shell. Try isuing it through inittab or through an at or cron job to get a more realistic environment if you suspect that. Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc A nicer trace would be done using 'set -x' . Add that line somewhere in the init script. Compare the output from the good case to the output from the bad case. BTW: any reason why the redhat and debian init.d script in the contrib directory still use LD_ASSUME_KERNEL=2.4.1 ? It is not used in the init.d script of the Debian package and not in the init.d scripts that appear to be newer in that directory. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /etc/init.d/asterisk barfing
On Thu, Sep 01, 2005 at 09:13:36AM -0600, Rich Adamson wrote: Not sure this applies, but there does seem to be a problem in the zaptel area that also impacts service... command. If I kill asterisk Do you manually kill the asterisk process? If so: what happens with the pid files and lock files? and try 'service zaptel stop', it now fails consistently as its unable to stop 'ztdynamic' (leaving zaptel still running since it can't unload). Why would you want to stop zaptel exactly? Zaptel is not some daemon like Asterisk. That init.d script loads/unloads some kernel modules. Normally there should be no reason to unload them. However, 'service zaptel start' does succeed on my FC3 cvs-head (from Aug 25th) and one TDM card installed. Any chace Asterisk is actually still alive and prevents the kernel modules from cleanly shuting down? The Debian init.d in the contrib directory is far from being robust Sorry, went of at a tangent (that's what you get for half reading emails I guess!) Ok, guess the easiest thing to do is to check in the contrib directory, diff your one against the redhat (are you running redhat?) Mine is the same and it works fine, maybe you're running an outdated init script. Still worth trying the strace against the service command, at least it'll give you an idea of what it's trying to do. One last thing, did it work and then stop working or is this a fresh install and it's never worked? Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 01 September 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /etc/init.d/asterisk barfing Hi Steve :) The problem is not with the asterisk command, nor with safe_asterisk but with the /etc/init.d/asterisk script if I manually run /etc/init.d/asterisk start all's ok if I manually run service asterisk start it says that it has started, but hasn't :) Julian Steve Hanselman wrote: Try doing an strace on it and seeing what the last section shows you. i.e. strace asterisk -vvvc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: 31 August 2005 22:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] /etc/init.d/asterisk barfing Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and /var/run/asterisk.pid both exist (even after the run). Can't find any reasons or errors for this not working - does anyone have any clue on where to start looking - I need * to automatically start on init. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
