[asterisk-users] Asterisk call forward for T1 incoming calls
Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] handset forwarding Diversion header cannot be set on Local channels
is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: 202 sip:202@192.168.1.46;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-0483 to 'Local/33@test' (thanks to SIP/202-0484) and not all Telco providers handle diversion header gracefully, some dont like to see 202 in header. i tried to set the sip header in target 33@test but asterisk see's this as local channel and wont do sip add header: WARNING[13584]: chan_sip.c:20562 func_header_read: This function can only be used on SIP channels. is there anyway around this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AppVoicemail overwrites voicemail.conf
yes, thanks you! On Sat, Mar 22, 2014 at 9:13 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote: looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. Right, use passwordlocation = spool, create a secret.conf for each mailbox, now when a user changes their password, secret.conf gets updated not voicemail.conf. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AppVoicemail overwrites voicemail.conf
We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AppVoicemail overwrites voicemail.conf
passwordlocatio seems to be related to vmsecret from voicemail.conf sample : ; passwordlocation=spooldir ; Usually the voicemail password (vmsecret) is stored in ; this configuration file. By setting this option you can ; specify where Asterisk should read/write the vmsecret. ; Supported options: ; voicemail.conf: ; This is the default option. The secret is read from ; and written to voicemail.conf (or users.conf). ; spooldir: ; The secret is stored in a separate file in the user's ; voicemail spool directory in a file named secret.conf. ; Please ensure that normal Linux users are not ; permitted to access Asterisk's spool directory as the ; secret is stored in plain text. If a secret is not ; found in this directory, the password in ; voicemail.conf (or users.conf) will be used. ; Note that this option does not affect password storage for ; realtime users, which are still stored in the realtime ; backend. but the issue i was explaining was voicemail.conf getting overwritten apparently by appvoicemail On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AppVoicemail overwrites voicemail.conf
looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. On Fri, Mar 21, 2014 at 9:31 PM, Al lists asteris...@gmail.com wrote: passwordlocatio seems to be related to vmsecret from voicemail.conf sample : ; passwordlocation=spooldir ; Usually the voicemail password (vmsecret) is stored in ; this configuration file. By setting this option you can ; specify where Asterisk should read/write the vmsecret. ; Supported options: ; voicemail.conf: ; This is the default option. The secret is read from ; and written to voicemail.conf (or users.conf). ; spooldir: ; The secret is stored in a separate file in the user's ; voicemail spool directory in a file named secret.conf. ; Please ensure that normal Linux users are not ; permitted to access Asterisk's spool directory as the ; secret is stored in plain text. If a secret is not ; found in this directory, the password in ; voicemail.conf (or users.conf) will be used. ; Note that this option does not affect password storage for ; realtime users, which are still stored in the realtime ; backend. but the issue i was explaining was voicemail.conf getting overwritten apparently by appvoicemail On Fri, Mar 21, 2014 at 5:36 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] is this expected behaviour?
i noticed in asterisk 10.12.3, i get messages like this: [2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63 but not mentioning attacker ip (to be used for fail2ban) is this expected? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Register DOS attack
I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Register DOS attack
Hi List Recently i have noticed this attack on couple of servers, usually a foreign IP starts sending tons of register request without any answer to authentication, if you type sip show channels in cli you will see tons of these: 1.2.3.4 (None) 2389603298 00101/1 0x0 (nothing)No Rx: REGISTER since there is no authentication in place, asterisk does not see any failed register attempt, so there wont be anything added to log file as failed attempt. thus fail2ban wont see any activity and wont block the IP. it simply brings down the internet link and the box due to too many sip channels. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo on Sangoma A400 and background noise
I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: As you mentioned you have tried Oslec algorithms for echo cancellation.Which is a good way to solve echo cancellation issues. If that is not woking for you you may want to upgrade to hardware echo cancellation..with cards which have echo cancellers. Hope this helps. -Sri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On 7/30/09, Steve Totaro stot...@asteriskhelpdesk.com wrote: The first time is always free :) On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load balancing Asterisk.
Foundry serverIron does support SIP and its ASIC not a linux box Load balancer like F5, Refer to Chapter 10 (page 677) of ServerIron manual. It explains everything in detail. Also you may need to play with source nat a little bit to make your specific configuration work, but it should work, at least in theory. On Thu, Nov 20, 2008 at 10:25 AM, Alex Balashov abalas...@evaristesys.comwrote: SIP wrote: As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-based SIP load balancing and only worked with certain SIP proxies. I assume that is because there is no way RFC-supported way to insert a cookie into a SIP session that persists throughout the entire exchange with a client, including all in-dialog requests, subsequent sessions, etc? The only way I know of to make a cookie stick on the UAC side is to put an LR parameter into the route set, but that will only last within a dialog. So, I'm assuming certain SIP proxies had proprietary ways of getting around that in order to work with F5? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP security
yes, make sure context line in general area has a dummy context, something with one line to hangup. On Fri, Nov 28, 2008 at 12:56 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Nov 28, 2008 at 11:00 AM, Mike l...@virtutel.ca wrote: I was looking at my CLI the other day, and found a lot of those types of messages: NOTICE[2242]: chan_sip.c:14383 handle_request_invite: Call from '' to extension '000452555169' rejected because extension not found. Looking at the IP, it originated from Asia and was clearly an attempt to screw with my Asterisk server. My quick fix was simply to block the IP adress at the firewall level. So that was the end of that. What I don`t get is how the person got that far. How could he attempt to dial extensions (even though he probably was in the default context which has nothing in it) when all my SIP peers are either password protected or linked to a fixed IP. And, more to the point, Call from `` means a call from what exactly? It's not one of my phones, it's not one of my peers…..Shouldn't the lack of a peer be enough to block the would-be hacker from tyring extensions? Any help is appreciate, I clearly don't understand SIP peers. Mike I think if you remove context from the [general] section, you would not see these messages. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk REFER
is this a feature in asterisk? On Mon, Sep 15, 2008 at 3:20 AM, Patrick Maartense [EMAIL PROTECTED]wrote: Ice is the feature you're looking for I think If two clients support ice, a direct link between them will be made -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Al lists *Sent:* Dienstag, 09. September 2008 23:40 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk REFER Hi All, from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ? No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.6.19/1661 - Release Date: 09.09.2008 04:58 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk REFER
Hi All,from what i'm understanding, Asterisk is back to back user agent. Base on this my initial thought was even if we enable reinvite in sip.conf, asterisk still will be in sip path after transfer. But i read some information in asterisk using refer to transfer a call completely to another sip or per say, a call comes in from voip provider and get transferred by asterisk to a cell phone number by using same provider and then asterisk will not be in SIP path anymore. is it doable ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with DTMF on IVRs
last time i had this issue with teliax, they recommended to upgrade to 1.4 On Fri, Aug 29, 2008 at 3:44 AM, Chris Mason [EMAIL PROTECTED] wrote: I tried DTMFmode=auto and it did not help. Any further ideas? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] finding out on hold channels
While this is in place, how about sip show channels and show channels ? On Fri, Jul 25, 2008 at 4:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Jul 25, 2008 at 2:59 AM, Al lists [EMAIL PROTECTED] wrote: I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. Hi, There was recently a thread regarding this on asterisk-dev (http://lists.digium.com/pipermail/asterisk-dev/2008-June/033466.html). There was message explaining how to do this by adding custom code to Asterisk sources, and I guess it could be already done in trunk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] finding out on hold channels
I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco vs Asterisk
I agree, No manager gets fired even if a Cisco Call Manager goes south. that's not the case with Asterisk. With limited experience that i have with both, i hit more bugs using Asterisk than a CCM, but this is not relevant to your final answer. If you can afford CCM, and you can live with less flexibility and features, i would choose Cisco. If you prefer to have cheaper solution and more features and flexibility, Asterisk is good. With Cisco, everything is cisco, handsets are designed for Cisco, it connects to Exchange much more in depth than even microsoft response point. unlike Asterisk, unfortunately exchange integration is not something you may get in close future and that can be a deal breaker for some companies, but you dont pay per seat license. and so on. On Thu, Jul 24, 2008 at 2:56 PM, Senad Jordanovic [EMAIL PROTECTED] wrote: T G wrote: I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and Telepresence systems I have two IP patents for the VoiP Lite protocols and have been designing and building OSS IPBXs for companies including Google going back to 2001. I'm not mentioning any of that to be jerk I mentioned it to say I'm as qualified as anyone to to compare the CCM and OSS servers. The only fair way to compare the two is a list of weights features, for example if cost is your biggest feature then OSS is better, if support is your biggest feature than Cisco wins. When a customer is comparing the costly (TCO) and best supported systems in the world with hundreds of thousands installed systems for the large global companies on the planted backed by 54,000 employees and over $25b in the bank vs, a FREE system with one layer of support maybe two layers of support, the features don't even come in the evaluation in my opinion. I once asked a manager why did you buy the CCM and he said no one ever got fired for buying Cisco if anything wrong, If push the OSS and it goes I could loose my job. I would get a list of the important features, because there is no answer to your question of which is better. What you mentioned above is mostly correct presuming you are referencing OSS being provided by an organisation with limited resources and perhaps limited experience in OS. Spin that into a perspective of a well organised company harvesting full potential of OS, adding its own proprietary software level allowing it to offer value products and EXCELLENT support, then I will strongly disagree with you. In particular where customer solution isn't just a solution, but rather its products and people becomes your business's communications partner. Senad www.bicomsystems.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
If you are trying to reject an IP address to connect to asterisk, there is no need to run iptables. Each SIP definition in sip.conf can have: deny=0.0.0.0/0.0.0.0 permit=192.168.135.1/255.255.255.0 just set these values and it wont accept anything from that IP. On Mon, Jul 7, 2008 at 7:37 PM, Dovid B [EMAIL PROTECTED] wrote: - Original Message - From: spectro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 01, 2008 8:02 PM Subject: Re: [asterisk-users] sip extension compromised,need help blocking brute force attempts On Tue, Jul 1, 2008 at 11:19 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Fix your logger.conf, then. -- Tzafrir Cohen What am I missing? [EMAIL PROTECTED] ~]# cat /etc/asterisk/logger.conf ; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [logfiles] ; ; Format is filename and then levels of debugging to be included: ;debug ;notice ;warning ;error ;verbose ; ; Special filename console represents the system console ; ;debug = debug ;console = notice,warning,error ;console = notice,warning,error,debug ;messages = notice,warning,error full = notice,warning,error,debug,verbose ;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 = notice,warning,error ; [EMAIL PROTECTED] ~]# The script seems to run off the messages log. Uncomment the messages line and the reload the logger in asterisk (logger reload from the CLI). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
i used it on one server a little while ago. my primary use was ability to show each user's status on spark. i did not get consistence results, phone status was not accurate. and did not try it after that, maybe its fixed in newer versions. On Fri, Jun 20, 2008 at 2:44 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: See below: Erik Anderson wrote: On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson [EMAIL PROTECTED] wrote: So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Increased data on the hard drive won't really have an affect on reliability or performance. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? I can't speak directly to the asterisk + openfire situation. I can, however, say that I've been running openfire for nearly a year now on a very highly-loaded server (other than openfire, it's running nagios and cacti, monitoring about 300 devices around our network) - the load average on this 5-year single processor old dell server is pegged near 1.00 24x7. I haven't had a single problem with openfire, and I have between 50 and 100 open sessions at any one time. In the year that I've been running openfire, I've only had to restart it once, and that was to upgrade the software. It takes very little CPU, and a modest amount of RAM. Is it better for production to run Openfire on a separate server than the PBX? What's your definition of better. Is it better to not have all your eggs in one basket? Is it better to only need to purchase one server? Is it better to only have one server to manage/update/etc versus two? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Speaking personally, I'd have no problems putting openfire and asterisk on the same box. If needed, you could even just nice the We run with the openfire process on the same box as the * server - we have not had a single problem with openfire in over 2 years now. openfire process down to a lower priority than asterisk - it's not as latency-sensitive as asterisk is. I'd doubt you'll need to do that, though. -Erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] suggestions for IAX ATA device or phone in US
anyone has used or bough one? would appreciate comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Drag and Drop transfer application
any of you guys have used FOP for drag and drop transfer on 30 40 phones environment? how stable is that? I'm playing with it but so far drag and dropping phone icon to another phone disconnectes the call. On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins [EMAIL PROTECTED] wrote: Al lists wrote: Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? Yes. Maestro Control Panel (I authored this one) http://www.datatrakpos.com/pos/datatalk/maestro.aspx. There is also the nice flash based Flash Operator Panel http://www.datatrakpos.com/pos/datatalk/maestro.aspx There a couple of other ones out there too that I thought were nice, but can't remember the names. You should be able to find them by gooling for Asterisk Control Panel or such query. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Drag and Drop transfer application
Hi list, Any good drag and drop transfer call application for windows based systems you can advise ? Something like HUD perhaps? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf wont load completely
I have seen this issue on both 1.2 and 1.4, was not able to reproduce to find a cause or bug. I have seen this after power failure boot up. show sip peer command shows most of peers, except one or two (in my cases trunk) . if i issue a sip reload command, it will show all of them. I can write a script to reload asterisk after a minute of boot up but i wanted to see if anyone else has seen this issue or has any thoughts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
I just wanted to check one more thing, system is connected to PSTN via SIP trunk ( No echo) , and terminates to customer analog phone's via Adit 600 fxs. I do not see any need for echo cancellation in this setup. There is no far end hybrid source, Any other thoughts? On Thu, Apr 3, 2008 at 8:18 AM, Darren Wright [EMAIL PROTECTED] wrote: I've used Adit600's almost exclusively for my installs. All have worked great for me. -D From: [EMAIL PROTECTED] on behalf of Steve Totaro Sent: Thu 4/3/2008 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks. Just Google Quintum Tenor AX. Well worth the money. Thanks, Steve Totaro On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote: Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX IP Phone
Atcom supports IAX: http://www.voip-info.org/wiki/view/AT-530 On Sat, Apr 5, 2008 at 11:17 AM, Joseph [EMAIL PROTECTED] wrote: On 04/05/08 05:16, bilal ghayyad wrote: Hi All; Till now I am not able to find a good IAX IP Phone or Gateway that can be used with good quality. Anyone can advise for good one? Regards Bilal I've not seen IAX phone so your best option will be IAXy adapter from digum. It works OK; but it is not free of bugs. -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Darren and Jerry, is it possible to post your config if its different than: http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check thank you! On Thu, Apr 3, 2008 at 8:18 AM, Darren Wright [EMAIL PROTECTED] wrote: I've used Adit600's almost exclusively for my installs. All have worked great for me. -D From: [EMAIL PROTECTED] on behalf of Steve Totaro Sent: Thu 4/3/2008 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks. Just Google Quintum Tenor AX. Well worth the money. Thanks, Steve Totaro On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote: Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web page to show online extensions?
you can see users status in Jaber, Install Open fire Jabber server with Asterisk pluging. On Thu, Apr 3, 2008 at 1:55 PM, Earl Terwilliger [EMAIL PROTECTED] wrote: On Thursday 03 April 2008 02:59:07 pm faraz wrote: FOP is quite clunky! one reason i wrote the event montor... which is in PHP (and Ajax or rather Ajap) and does not poll the asterisk manager (which in my opinion overloads asterisk) Also the flash is almost un-usable with a large number of extensions Would love to see something in PHP/Ajax which could be lightweight and fast. We are working on something along those lines which we should be able to release in a few months. On Thu, 2008-04-03 at 14:42 -0400, Dean Collins wrote: Cute :) I was thinking about getting something more complex developed but yes FOP is a great product though getting a little old.time for the next version? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, 3 April 2008 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Web page to show online extensions? On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote: What about building an Adobe AIR application that can do this. Any application that connects to the manager interface can do that. It can be AIR, or FIRE or GROUND. The FOP exists and does that. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Its Nice, i agree, but we are looking at $4k to $5k with this. On Wed, Apr 2, 2008 at 1:17 PM, Andrew Latham [EMAIL PROTECTED] wrote: Here I will say it http://xorcom.com On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote: I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] TuxTone Inc. http://www.TuxTone.com */ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Bad memories from AudioCodec :) On Wed, Apr 2, 2008 at 7:48 PM, Edwin Lam [EMAIL PROTECTED] wrote: Andrew Latham wrote: Here I will say it http://xorcom.com alternatively: http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote: I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to wait before sending DTMF in DIAL command
If you are asking about dial command on analog lines, here is what i do :
exten = _NXX,1,Dial(ZAP/g1/ww${EXTEN})
that should give you 2 seconds before actually start dialing, its good way
to wait for analog lines to stabilize first before dialing.
On Tue, Apr 1, 2008 at 9:49 PM, Pete Kay [EMAIL PROTECTED] wrote:
Hi friends,
Is there anyway to have Asterisk to wait for 1 second before sending a
DTMF using the D() option?
Thanks for your suggestion.
Pete
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[asterisk-users] Need some input for Quad T1 and channel banks.
I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Im guessing T1cas not PRI,just because its giving 24 fxs per T1. Steve, what are my options for SIP to fxs? thank you! On 3/31/08, Doug Lytle [EMAIL PROTECTED] wrote: Don Pobanz wrote: Doug Lytle wrote on Monday, March 31, 2008 5:40 PM This does not sound right. If it is 2 PRIs then it should be 46 channels I may have the terminology incorrect. I don't have a D channel, so I guess this would be called a T1 then? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got SIP response 406 Not Acceptable
I'm getting Got SIP response 406 Not Acceptable back from 10.0.1.2 occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Got SIP response 406 Not Acceptable
Nope, Coded is Ulaw on both sides and also this issue happens occasionally with no change. On Wed, Mar 26, 2008 at 6:17 PM, Adrià Vidal [EMAIL PROTECTED] wrote: Seems a codec problem, check the sip.conf from that spa942 On Wed, Mar 26, 2008 at 11:59 PM, Al lists [EMAIL PROTECTED] wrote: I'm getting Got SIP response 406 Not Acceptable back from 10.0.1.2 occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail Server
Or maybe you can show him some links ;) Try this for send mail: http://docs.snake.de/smtp-auth.html this is very common these days and to make it more fun each mailserver (provider) has their own criteria to decide if your email is spam or not. to give you and example: make sure you are using static public IP address for outgoing mails, have a PTR record for that IP and also A record for the fqn that those mails are coming from. For smtp auth you need to have saslauth in place and most recent sendmails are compile with saslauth these days. I did not have 100% success with smtp and sasl and i believe that was caused due to have different TLS versions. anyway that link should put you in the right direction and if anyone else has a better/easier mta that handles smarthost and auth flawlessly, please comment. On Sun, Mar 16, 2008 at 3:48 PM, linuxian iandsd [EMAIL PROTECTED] wrote: well, maybe ou're on the wrong list (talkin sendmail in an asterisk list !!!) you're better in sendmail's list. anyway, you need to modify sendmail.cf file, just a few tweaks it will be ok. you will need a smarthost, what is a smarthost ? thats an smtp server that is allowed to send mail to the world, without it you can't send mail, this smarthost will be your isp's smtp server noone else's unless you know a lot of ppl around. otherwise your mails will get nowhere. if you need an sendmail.cf file example i can paste it for you here. also dovecot.conf will be valuable for you. hope this helps. On Fri, Mar 14, 2008 at 1:52 PM, Felipe Trevisan [EMAIL PROTECTED] wrote: How would you relay on Google Apps, as Google requires SSL or TLS authentication? How can I configure sendmail to do this? Actually, sendmail is trying to send email directly, and I get the response below. I´ll now try Mike Hammett´s solution. Thanks, Felipe Trevisan *Message contents* The original message was received at Thu, 13 Mar 2008 23:49:31 -0300 from trixbox1.localdomain [127.0.0.1] - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 550-5.7.1 [201.6.192.115] The IP you're using to send email is not authorized ) - Transcript of session follows - ... while talking to gmail-smtp-in.l.google.com.: DATA 550-5.7.1 [201.6.192.115] The IP you're using to send email is not authorized 550-5.7.1 to send email directly to our servers. Please use 550 5.7.1 the SMTP relay at your service provider instead. a44si4966479rne.2 554 5.0.0 Service unavailable *Failed delivery status* *Final recipient* [EMAIL PROTECTED] *Reason for failure* 550-5.7.1 [201.6.192.115] The IP you're using to send email is not authorized *Remote mail server* gmail-smtp-in.l.google.com *Reporting mail server* trixbox1.localdomain On Thu, Mar 13, 2008 at 7:13 PM, Mike Hammett [EMAIL PROTECTED] wrote: Through help from people on the lists and then further investigation based on those results, here is what I did. 1) I set the office to a statically assigned IP instead of from the pool. 2) I made an A entry on one of my domains aiur.ics-il.net (where aiur is the machine name). 3) I added aiur.ics-il.net directly after 127.0.0.1 in the /etc/hosts file (copied below). 4) I set the from email address (serveremail) in /etc/asterisk/voicemail.conf to something at the domain I created ( [EMAIL PROTECTED]). 5) Presto! [EMAIL PROTECTED] ~]# cat /etc/hosts # Do not remove the following line, or various programs # that require network functionality will fail. 127.0.0.1 aiur.ics-il.net Aiurlocalhost.localdomain localhost ::1 localhost6.localdomain6 localhost6 -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - *From:* Mike Hammett [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Thursday, March 13, 2008 4:04 PM *Subject:* [asterisk-users] Mail Server I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?
Simple, add new interface in your system and put BOOTPROTO=dhcp in ifcfg-eth1 if you have one gateway you can add that in the same file or in /etc/sysconfig/network, or if you have multiple gateways, you need to define a route to your voip service through that interface. On Tue, Mar 18, 2008 at 7:29 AM, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I'm about to test VOIP connection (from my ISP provider) directly through dedicated network card instead of going through ADSL gateway with analog phone port - SPA 3000 - Asterisk. I need to have eth2 set on dhcp (to retrieve IP automatically) and then work with it under Asterisk as dedicated VOIP trunk. Anyone with more insight how to setup such situation ? Any more info anywhere ? Thanks in advance, regards, Bulek. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Avaya 4610 handset
i was reading posts on wiki and noticed lots of posts about Avaya 4610 handset having issue with MWI, Anyone has any more updates? Is this still the case? Any good tutorial for configuring these phones and Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Nagios
Has anyone checked asterisk with check_udp plug in? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HPEC
Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
Always rely on free -m to see how much free memory you have not top. in terms of memory leak, i have asterisk running on servers with uptime of 400 days (CentOs), if there was any leak, i'm guessing i would have crashed server long time ago. On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey [EMAIL PROTECTED] wrote: If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF / Speed Dial
check here: http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce On Feb 6, 2008 4:22 PM, Tim Nelson [EMAIL PROTECTED] wrote: Could you possibly post what steps you took to make this work so others (including myself :-) ) may benefit? Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Michael Munger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 6, 2008 5:05:20 PM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Polycom BLF / Speed Dial I figured it out. Thanks anyway! Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Munger Sent: Wednesday, February 06, 2008 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom BLF / Speed Dial Is there a way to configure the buttons on the phone that are normally reserved for line registrations so that I can do a one-button pickup of a parked call complete with Presence? The goal is to have a couple of the line registration buttons show me who is on park orbits 701 and 702 so that I can pick them up with one-touch. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switch QOS requirements
Very Nice! Its much more reliable than translating DSCP to COS by switch which i'm not sure which switch does that and which one doesn't, and how they do it considering some gray area when you translate from DSCP to COS. On Feb 4, 2008 5:26 PM, Jared Smith [EMAIL PROTECTED] wrote: On Sun, 2008-02-03 at 22:42 -0700, Al lists wrote: Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. In Asterisk 1.6, you will be able to set both the COS and TOS values. The sample sip.conf in the Asterisk 1.6 betas contains the following, to show you just how much you can adjust things :-) ;tos_sip=cs3; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5; Sets 802.1p priority for RTP audio packets. ;cos_video=4; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switch QOS requirements
Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. I guess to be safe, you would need to create 2 VLANS and in the switch define on VLAN as a high priority VLAN. On Feb 3, 2008 7:06 PM, Michael Graves [EMAIL PROTECTED] wrote: On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote: John Williams [EMAIL PROTECTED] writes: We are tearing out legacy PBX and replacing with Asterisk PBX and new LAN for our 90+ person operation. Question: what QOS capabilities (protocols, etc) does Asterisk support/require in a LAN switch to deliver business grade phone service? Thanks If you have one switch for the whole network, you're generally fine without QoS. Switches these days can handle full bandwidth on all ports at the same time. Anyway, Asterisk is no different from other PBX's when it comes to QoS. Should it turn out that you actually need it on the LAN, just be sure you set the tos parameters in sip.conf to something that is prioritized by the switch. It tends to be more of an issue when you're sending calls over a link with limited bandwidth. Usually more of a concern in the router. Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking lot
Is there any way to have Asterisk call an extension in dial plan instead of original extension after timeout? Like extension A puts the caller in parking lot, he leaves the phone and forgets about it, instead of having that phone rings after timeout, have a group of phones rings. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk optimalization
I have been using Dell servers and have no issues with linux, in fact when i implemented my last install with their top of the line server (dual xeon quad core and SAS drives on Perc 6i) i was amazed how smoothly it went trough. Beside that i like their open manage, it runs nice on linux and its a handy tool on remote location servers. On Jan 24, 2008 7:55 AM, Daniel Guthrie [EMAIL PROTECTED] wrote: That's funny… I seem to remember installed Deb/* on a Poweredge 2950… …must be slowly losing my mind. Another side effect of using Asterisk? Dementia? *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Gopal krishnan *Sent:* Thursday, January 24, 2008 2:51 AM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] asterisk optimalization Hi, Dell is not a recomeded server for linux. Its only compatible with windows. On Jan 24, 2008 12:02 PM, Goke Aruna [EMAIL PROTECTED] wrote: ram wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm That result is suprising! but i have DELL 2950 with 2 X 3.0GHz CPU on 6GB ram, equiped with 8e1 link (2 sangoma A104D) running FC5. I installed chan_ss7-1.0 with asterisk-1.2.25 doing transcoding, and each time calls get to 120+ the cpu is fully utilized. the calls come from sip to the ss7 link. can someone advice me on what I can do to improve the performance. goksie NB. I felt we re talking on the same topic thats why i added my own experience. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Thank you Paul! Its impressive! On Jan 23, 2008 4:55 PM, Paul Hales [EMAIL PROTECTED] wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm It was the bottom news item on voip-info.org - I was worried I would have to really search for it! later, PaulH On Wed, 2008-01-23 at 18:30 -0500, Steve Totaro wrote: Link? Thanks,Steve Totaro On Jan 23, 2008 6:08 PM, Paul Hales [EMAIL PROTECTED] wrote: There was a cool paper written a a few months ago where they tested some older dell servers - full details of specs and tests were available. PaulH On Thu, 2008-01-24 at 08:54 +1100, Daniel Cole wrote: Sorry to be a little OT.. But may I ask what some more of the specs are for that machine? Just trying to get an idea of what different hardware can achieve. Thanks,Daniel __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Thursday, 24 January 2008 7:57 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk scalability I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Asterisk is highly multi-threaded and definitely takes advantage of multiple cores.There are a few places where concurrency could be further improved, but its really quite good in 1.4. (IAX in 1.4 does handle traffic using a thread pool so will take advantage of multiple cores).By the way, I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls. Steve ___-- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom-SIP response 500
Yes, this prompt will shows up on SIP 2.2.2 as well. I never had any issues with this though, it will clear up after next registration of phone. I just downloaded SIP 3.0 and have not got a chance to check and see if it happens with this firmware as well. On Jan 22, 2008 2:53 PM, Steve Johnson [EMAIL PROTECTED] wrote: I have just retested and agree that this error eventually does clear itself. However, in this test it took about 35 minutes and each Polycom phone produced between 1000 and 1300 error message lines at 1 to 0 second intervals (which I captured to the debug log). Once one phone starts flagging an error, all Polycom phones that are buddy-watching join right in. I triggered the problem by simply restarting asterisk: /usr/sbin/asterisk -rx restart when convenient. Sometimes (but not all the time) it will also start if you reload. All suggestions appreciated. S. On Jan 22, 2008 9:23 AM, Steve Johnson [EMAIL PROTECTED] wrote: I am using Polycom's SIP 2.2.0047 (the current release) and am seeing this. It seems to occur less often with extensions reload rather than just reload, but it would be nice to fix this. Tx. On Jan 22, 2008 8:30 AM, Steve Davies [EMAIL PROTECTED] wrote: On 1/22/08, Steve Johnson [EMAIL PROTECTED] wrote: Hi list, There are many Polycom experts on this list -- hopefully someone has a solution. With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk causes the Polycom 601 phones to start dumping these messages to the CLI. -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.2.x [snip] I have not seen this problem here since upgrading to 2.1.2 firmware. Or perhaps it was 2.2.0, one or the other. The phones now seem to recover on thier own when Asterisk returns. Cheers, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Register source port
Hello all, is there any way to tell asterisk what port to use for source of any registration request? for example the simple register command, register = user:[EMAIL PROTECTED]:port will send the register packet from asterisk_IP:5060 to proxy:port . Is there anyway to have asterisk to use different port instead of 5060 for each register command, like 5060 for the first 5070 for second .. ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK cd-rom
Cool! I didnt know Fedora has Asterisk in their repository. Nice! On Jan 5, 2008 4:26 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Jan 05, 2008 at 03:15:13PM +0530, Bhrugu Mehta wrote: hi, all i want to create cd-rom with asterisk. how it possible. when i put disk in cdrom it boot automatifcally and auto-start installation like TRIXBOX. any idea. Trixbox (Trixbox CE, actually) doesn't buid Asterisk on autostart. It uses Asterisk from packages. This is the only sane way to manage a release for more than one or two versions. Packages of Asterisk are included in at least the following distributions: Debian, Fedora, Gentoo, OpenSUSE and Ubuntu. Each of them has its own way for creating a custom install CD with some extra packages. With Debian I was able to do that with just minimal changes to the installer (just a preseed file), and my own repository of backported packages. At least for the case of network installation. Alternatively look for DebianCustom. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I'm not looking at T.38 , at this time its terminating a SIP trunk with multiple DID's for fax. I'm using this configuration with linksys PAP ATA and satisfied with results. I'm looking at removing these ATA 's and using Asterisk ( or giving it a try ) for terminating fax. Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve The most recent versions of t38modem can apparently provide both a SIP and H.323 T.38 to class 1 FAX modem interface for HylaFAX. What it cannot provide is an audio FAX interface. The sipmodem code I am working on will integrate audio and T.38 FAX processing in a single SIP entity. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Guys! what i was looking here was a simple hint/recommendation for installing IaxModem and Hylafax. Let me try it myself and see how feasible this solutions is. On Jan 1, 2008 5:02 PM, Steve Underwood [EMAIL PROTECTED] wrote: Jonn R Taylor wrote: I have always said that if some one said it can't be done, they did not try hard enough. FYI... I love this. Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. As the person behind the tools you are relying on, I can say you haven't tried hard at all. You are just lucky, and almost certainly just being very reliant on the majority of your FAXes using ECM mode, and retrying a lot. Trying hard for FAX over IP means implementing T.37, or at least T.38. These are engineered solutions, not pot luck. Your present arrangement assumes G.711 (not available a lot of the time), no signal manipulation in the system beyond your controls (getting rarer and rarer), a very crude network doing nothing to improve voice quality (should be getting rarer too), limited packet loss (which is truly pot luck over the internet, which you say you use), and a few other magic qualities. A number of people claim solid FAXing results across VoIP paths, like they've achieved some engineering breakthrough. The claims tend to evaporate under closer inspection. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
at this time is terminating a SIP trunk, each DID will get its own fax box. I guess at this time i'm looking to find a tutorial for installing iaxmodem and hylafax as it seems to be the answer. On Dec 31, 2007 9:11 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. What are you trying to do and do you have a T1 or ISDN line? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Any recommended how to for 1.4 iaxmodem and hylafax+ ? On Dec 29, 2007 6:49 AM, Doug Lytle [EMAIL PROTECTED] wrote: Al lists wrote: So HylaFax and IaxModem is more preferred than using rxfax/txfax ? any reason? HylaFAX+ has built-in support for handling transmission errors. It used to be (Not sure now) that when rx/txfax encountered an error, it just quit. HylaFAX+ also has many features available for converting/scheduling and routing inbound/outbound faxes. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
thank you all, still i'm seeking answer to original question, which one is more preferred in fax servers with 100 usres? On Dec 29, 2007 12:10 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Dec 29, 2007 at 08:43:30AM -0700, Al lists wrote: Any recommended how to for 1.4 iaxmodem and hylafax+ ? Just a note: the interface of iaxmodem to Asterisk hasn't changed since 1.2 . Chances are older documentation will be relevant. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Fax
what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
So HylaFax and IaxModem is more preferred than using rxfax/txfax ? any reason? On Dec 28, 2007 6:40 PM, Lee Howard [EMAIL PROTECTED] wrote: Al lists wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. I think that you mean to say HylaFAX and IAXmodem or txfax/rxfax ... because spandsp is but a DSP/DCE library, and it cannot work alone, and iaxmodem uses spandsp. Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT: Best Speakerphone
Agreed! Polycom and Polycom and Polycom!! On Aug 20, 2007 3:26 PM, Michael Graves [EMAIL PROTECTED] wrote: Sorry to top-post..but I haft agree here. Polycom is the KING of this sort of thing. Also, there really is a difference beteen a desk phone and a conference/borard room phone. Having multiple mics, aimed in different directions makes the dedicated conference phone a much better device hen people are all around a table. Some even support remote mics so that your can mic the ends of the table while the main phone sits in the center. Further, I do have Aastra 480i's and various Polycom desksets. They're great. Good quality speakerphones on both, but the conference phone is truly orth the price. Michael On Mon, 26 Nov 2007 15:28:49 -0500, asterisk wrote: With out any question POLYCOM! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 26, 2007 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Semi-OT: Best Speakerphone We're using Aastra 480i phones, and their speakerphone is great. I even have one in our datacenter and the speakerphone is usable even with all the noise of the server fans. I also have a great contact if you happen to be in Canada wanting to buy some. Bob On Mon, 2007-11-26 at 12:30 -0700, Ken Williams wrote: I'm looking for recommendations on speakerphones for a conference room. We're using Grandstream GXP-2000 which we've been very happy with on all accounts, except the speaker phone. Speaker phones on these units are extremely bad, picking up any and all background as well as having full-duplex issues (that is, when the other end is talking you can't talk over them to interrupt or whatever). So, before I go buy random phones for testing I thought I'd get some recommendations. Thanks, Ken ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf and it's impact on CLI
this message is basically tells you asterisk is not running. can you check and see if asterisk is running and present in memory? something like ps -ef | grep asterisk On 10/20/07, Dominic Son [EMAIL PROTECTED] wrote: I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)' Basically, the difference between 'asterisk.conf' file is as follows: v 1.2.9 (installed through trixbox) astrundir = /var/run/asterisk v 1.4.4 astrundir = /var/run So in my new servers, if i keep it as '/var/run/asterisk, my DID phone will work with stanaphone (in which i'm crapping in my pants if they'll exist cause they never return emails). Though CLI won't work. if i do '/var/run', my DID won't work, but CLI will... I've tried just coping over the extensions_additional.conf and sip_additional.conf files from my old setup to my new one, and that didn't work. Maybe I should just install my previous version. Are there QoS differences though? I'd rather not regress if that were the case. -- Anything else, let me know. - Dominic It is not the force of a stroke that makes fine art ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom ip330/ip501 second ethernet port
I Just wanted to add something here, Having separate VLAN does nothing in terms of QOS. In fact having a computer feeding from phone make more sense because phone will untag packets coming from PC. and after that its all about your switch how to prioritize packets. Unless there is a way in your switch to prioritize one Vlan over another Vlan, ( i guess it depends on your manufacture, i think Cisco does that and also uses CDP to discover phones) Having different Vlans is not your answer. the most you get is less broadcast. On 10/19/07, David Gomillion [EMAIL PROTECTED] wrote: On 10/19/07, Kevin Smith [EMAIL PROTECTED] wrote: Robert McNaught wrote: Hi, Has anyone had any great difficulties with QoS using the second ethernet phone in these Polycom phones for desktop machines in a converged network? I had heard that these can cause difficulties when used in this manner. I have always tried to persuade customers to go with 2 ethernet drops per workstation to avoid having to use the phone as a switch. I apologize for this question not being directly related to asterisk, but since Polycom phones are used a lot with asterisk, it seems a good place to post ;-) Robert McNaught Hi Robert, While I'm not sure how our network compares with yours, we run about twenty 601 phones along with our office workstations (some stations are without a phone). Each station with a phone is connected with the other Ethernet port on the phone so we have one drop to each station. The phones are on a separate VLAN from the rest of the network as well. From the user end, I have not had a report of any problems with the connections, call quality, etc. I would say give it a shot, maybe with a larger network that could change, but for a small office like I'm in charge of, it is working just fine. Kevin We have a medium-sized network (120 polycoms of various persuasions, and 80 workstations), and we haven't had any real problems with phones ruining QoS. We have the phones on separate VLANs than the workstations. Actually, every switch has 4 VLANs defined: 2 voice, 2 data, so no VLAN has more than about 12 devices (about because sometimes we have to put a pocket switch in a room where the people want to add yet another computer). The echo from SIP to SIP with people using cheap headsets has affected us far more than any problems with PCs trying to suck the bandwidth. If I remember correctly, recent firmwares on the Polycom phones pretty much do the right thing, giving priority to the phone traffic. To summarize: works OK for us. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
i'm using Polycom 601 in an office of 30 handsets. I have not heard my customer complaining about phones being rebooted after page. On 10/9/07, Bill Andersen [EMAIL PROTECTED] wrote: I could not tell you in asterisknow but I use this feature with Polycom phones on all of my installs. It is very well documented in voip-info.org Do you have any problem with the Paging when there are say 20 phones in the page group? We have a IP601 that is used by the receptionist and has 2 side cars. We have to keep presence (Buddy List) enabled so the sidecar lights go on and off. However, about 1 out of 10 times the receptionist pages, her phone reboots. Polycom says it can't handle the traffic from the buddy list presence notifications. Have you seen this? Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 and G723 and how to install it
Not sure about 723 bu you can buy 729 from digium just got to their website and its really easy to install, it comes with all instructions you need. On 10/7/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; From where I can buy the G.729 and G.723 licenses, and how I can install it on Asterisk so I can use it? Anyhelp? Regards Bilal Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos. http://autos.yahoo.com/index.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oddball time problem in CID
check tz option in your voicemail.conf On 10/5/07, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours off (early)??? I am really confused where could CID be getting this bogus info??? I am using Centos 4.5, Asterisk 1.2.7.1 and Freepbx version 2.3.0.3 Thanks Chuck Bunn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
Here is how i overcome this problem,
ignorpat = 9
exten = 9*,1,Dial(ZAP/1/w)
press 9* from your handset and after 1 second you have POTS line dial tone
on your phone,
On 10/3/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
It would be ugly, but you could prefix a zap channel or group number
before the phone number to dial. Using groups for an example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group 4...
bilal ghayyad wrote:
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone, not a dialtone provided by Asterisk.
Al lists wrote:
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be possible to assign an
dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be
via
that specific line.
Just assign a key on your phone to dial that
extension, and you will
have
dial tone on selected line,
then as a traditional PBX you can send any digits to
your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
ignorepat continues dialtone after a leading digit
has been dialed
on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
Dear List;
How can I place a call via Zap/g1 (group) but
need to
determine the line (FXO port)
that will go via it?
Simply don't use groups. Use channels directly.
To dial via the
specific
Zaptel channel NN, use Zap/NN
Am I missing anything?
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's
updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
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Re: [asterisk-users] What's the deal with ATAcomm?
Send me an email off the list, i have em somewhere in my HDD. On 10/2/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Kenneth Padgett wrote: Dear Atacomm Customers, We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm and its parent company Ataractic Corporation has ceased operations. We appreciate the 7 years of loyalty and support from our customers. We sincerely regret any adverse effects this may have caused. Well that just stinks. Anyone know how I'm going to get the latest Polycom firmware for all the phones I bought from them?!? I wish Polycom would realize the companies you buy stuff from sometimes go under. You can get the 2nd to the latest from Polycom without having to go thru a reseller. I don't know if that is acceptable to you or not. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting a specific line from Zap/g
ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting a specific line from Zap/g
Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
yea thats what i did i put SIP 1.6 and its working like a champ, there should be a way to get it working with 2.2, i'll wait for my next 601 and play with it. On 9/26/07, Doug [EMAIL PROTECTED] wrote: At 00:18 9/26/2007, Al lists wrote: One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. I guess they still haven't fixed that. The 601 that we have is using: 1.6.7.0098 On 9/25/07, Mike mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld ), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'http://somedomain.comsomedomain.com'(http://66.16.26.106 66.16.26.106) TTL=83485 copy |3|00|' ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(http://66.16.26.10666.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(http://66.16.26.10666.16.26.106 )' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. On 9/25/07, Mike [EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'somedomain.com'(66.16.26.106) TTL=83485 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stanaphone issues. can someone verify my config?
any firewall in between? On 9/18/07, Richard [EMAIL PROTECTED] wrote: Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related to broken sip phones, not a voip server. Alternativly, since it seems that stanaphone is a bit of a hit and miss from some other reading, is there any other functional US inwards provider for free that doesn't need a credit card that works well with asterisk? The softphone works, but I really need to get it going to my phones in the house instead. Soft client was closed when testing the asterisk. Many thanks. Richard Malcolm-Smith... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallWithUs Service?
Actually Cbeyond does that and their quality of voice is much better analog lines. On 9/15/07, Steve Totaro [EMAIL PROTECTED] wrote: Your best bet is to get your VoIP service through whoever your ISP is. If Global Crossing offered cheap VoIP (in comparision to some of their TDM offerings), I would consider it. It's all IP in the core now anyways, no real reason to use TDM for the last mile. Maybe it has something to do with the number of simultaneous calls you can stuff down a data T1 using G729 Thanks, Steve Al lists wrote: In VOIP, your quality of your voice is as good as your network. if you want clear call quality, QOS is a must. Well, when the call leaves your network and enters internet, QOS is not enforced. As a general rule choose the closest to your network. for me its Teliax, i get to their proxy after 7 hops. On 9/14/07, *Anthony Messina* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote: I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? it was your message that prompted me to take a look at callwithus.com http://callwithus.com. i currently use diamondcard.us http://diamondcard.us (via iax2) and have had only 2 issues in 9 months where some calls to verizon cell phones would get a congestion signal if they didn't answer instead of going to their voicemail. i called diamondcard and they fixed the trunk issue in a matter of an hour. call quality is decent. after signing up with callwithus.com http://callwithus.com, i find the call quality to be the same as diamondcard, though diamondcard bills in 30sec increments at 1.7 cents/min in the us and callwithus bills in 1 minute increments at 1.4 cents/min in the us. callwithus also has this thing where if you add a *31 to the number, it will choose their cheapest route. i'd say they are worth trying, so is diamondcard.us http://diamondcard.us. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery http://messinet.com/%7Eamessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISA and DTMF detection problem w/ FXO port on a TDM400
i did have same issue with DISA in 1.4 and TDM400 FXO,
I switched back to Authenticate and waitexten.
On 9/14/07, Benjamin M. [EMAIL PROTECTED] wrote:
Originally posted at http://forums.digium.com/viewtopic.php?t=18045
Hi!
I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing
DISA seems to prevent any DTMF detection capability when using the FXO
port of the TDM400.
Below, config A and B and their debug logs.
In Config A I use Authenticate() instead of using DISA password since it
demonstrates that it's DISA that seems to prevent DTMF detection when
using Zap/1. Otherwise DISA works flawlessly when calls are coming from
FXS port (TDM400), IAX, SIP channels and we have absolutely not
other problem detecting DTMF that we are aware of...
I see no active bug related to DISA at bugs.digium.com...
Any idea?
Ben.
*Code:*
---
zapata.conf
---
context=inbound-pstn
signalling=fxs_ks
rxgain=10
txgain=3
language=fr
channel = 1
I have tried to change gains without any result ...
(http://forums.digium.com/viewtopic.php?t=17769highlight=disa+dtmf)
; --- Config A --- ;
*Code:*
exten = 111,1,Answer
exten = 111,n,Authenticate(111)
exten = 111,n,DISA(no-password|internal)
; --- Dial sequence --- ;
*Code:*
PSTN line - TDM400
enter extension 111 - dial tone
enter password 111 - new dial tone
enter extension - I still getting the dial tone whatever I'm entering
timeout.
Here the debug log:
*Code:*
snip
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF end accepted without begin '1' on Zap/1-1
DTMF end passthrough '1' on Zap/1-1
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Oooh, got something to jump out with ('1')!
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
== CDR updated on Zap/1-1
Launching 'Answer'
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack
Launching 'Authenticate'
-- Executing [EMAIL PROTECTED]:2] Authenticate(Zap/1-1, 111) in new
stack
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'agent-pass' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: 1 on Zap/1-1
DTMF end '1' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '1' with duration 100 queued on Zap/1-1
DTMF end emulation of '1' queued on Zap/1-1
DTMF digit: # on Zap/1-1
DTMF end '#' received on Zap/1-1, duration 0 ms
DTMF begin emulation of '#' with duration 100 queued on Zap/1-1
DTMF end emulation of '#' queued on Zap/1-1
Set channel Zap/1-1 to write format gsm
Scheduling timer at 160 sample intervals
-- Zap/1-1 Playing 'auth-thankyou' (language 'fr')
Scheduling timer at 0 sample intervals
Scheduling timer at 0 sample intervals
Set channel Zap/1-1 to write format ulaw
Launching 'DISA'
-- Executing [EMAIL PROTECTED]:3] DISA(Zap/1-1,
no-password|internal) in new stack
Digittimeout: 3000
Responsetimeout: 1
Mailbox:
Context: internal
DISA no-password login success
Set channel Zap/1-1 to write format slin
Scheduling timer at 160 sample intervals
Scheduling timer at 0 sample intervals
[ asterisk isn't detecting any DTMF... -- ]
DISA extension entry timeout on chan Zap/1-1
Requested indication 8 on channel Zap/1-1
Set channel Zap/1-1 to write format ulaw
Scheduling timer at 0 sample intervals
Spawn extension (compagnie,111,3) exited non-zero on 'Zap/1-1'
== Spawn extension (compagnie, 111, 3) exited non-zero on 'Zap/1-1'
Soft-Hanging up channel 'Zap/1-1'
Hanging up channel 'Zap/1-1'
zt_hangup(Zap/1-1)
Hangup: channel: 1 index = 0, normal = 7, callwait = -1, thirdcall = -1
disabled echo cancellation on channel 1
Set option TDD MODE, value: OFF(0) on Zap/1-1
Updated conferencing on 1, with 0 conference users
-- Hungup 'Zap/1-1'
snip
; --- Config B --- ;
*Code:*
exten = 111,1,Answer
exten =
Re: [asterisk-users] CallWithUs Service?
In VOIP, your quality of your voice is as good as your network. if you want clear call quality, QOS is a must. Well, when the call leaves your network and enters internet, QOS is not enforced. As a general rule choose the closest to your network. for me its Teliax, i get to their proxy after 7 hops. On 9/14/07, Anthony Messina [EMAIL PROTECTED] wrote: On Thursday 13 September 2007 02:32:52 pm John Meksavan wrote: I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? it was your message that prompted me to take a look at callwithus.com. i currently use diamondcard.us (via iax2) and have had only 2 issues in 9 months where some calls to verizon cell phones would get a congestion signal if they didn't answer instead of going to their voicemail. i called diamondcard and they fixed the trunk issue in a matter of an hour. call quality is decent. after signing up with callwithus.com, i find the call quality to be the same as diamondcard, though diamondcard bills in 30sec increments at 1.7cents/min in the us and callwithus bills in 1 minute increments at 1.4 cents/min in the us. callwithus also has this thing where if you add a *31 to the number, it will choose their cheapest route. i'd say they are worth trying, so is diamondcard.us. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
I'm using Linksys Wip300 and i'm not happy with it. On 9/13/07, Dave Walker [EMAIL PROTECTED] wrote: On Thu, 2007-09-13 at 18:05 -0600, Stephen Bosch wrote: Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. S450: A recent firmware (few days old) upgrade seems to have solved the issue of being able to transfer calls. The handset still does not support 'Message Waiting Indicator, but does show missed calls. I am using this model, the audio IMO is superb and would recommend it. Failing that, there is the Aastra 480i-CT, (which is designed for the US market), but this includes a normal deskphone. If this as good as the other Aastra products, then you can't go too far wrong. Kind Regards, Dave Walker ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
Looks good! i need to find a distributer to buy one. On 9/13/07, Stephen Bosch [EMAIL PROTECTED] wrote: Anthony Francis wrote: Aastra now makes a full SIP DECT system with cell style seamless hand off from access point to access point. Caveat: This does not use standard wireless access points, you must purchase their access points and handsets. That's okay, it's a DECT phone. It's not supposed to use standard wireless access points. I'll look into it. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)
I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro [EMAIL PROTECTED] wrote: http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco UC 500
I'm trying to get some more information on this myself as its a new product from Cisco. What i know, Cisco attendant console works with skinny,Cisco page and SLA also works wiht skinny and not SIP. So its either having these or SIP. On 9/10/07, Drew Gibson [EMAIL PROTECTED] wrote: Jeremy Mann wrote: Is the Cisco UC 500 able to integrate with Asterisk? Specifically does it work via SIP? Just curious, as the Cold Call Cisco sales rep had no clue what SIP even was, and this device looks interesting. Google cisco UC500, hit #2 = http://www.cisco.com/en/US/products/ps7293/products_data_sheet0900aecd8061fb06.html Quotes: Core components of the Cisco Unified Communications 500 Series include: Cisco Unified IP phones, including wireless handsets and Session Initiation Protocol (SIP) phones PSTN interfaces and features: SIP trunks and RFC 2833 support Does that help? I'll bet Asterisk is cheaper though. :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Partitioning DSL input
Although you can find a router with QOS or dedicated bandwidth feature, I would suggest a QOS enabled Switch. Any IEEE802.1p enables switch,(these days less than $100 for 16 port) can do the job. you cant do alot when your traffic reaches internet, thats why most you can do is up to your modem. cos bit works best at layer 2 , and pretty much TOS is useless if you dont own your wlan line. On 9/10/07, David Gomillion [EMAIL PROTECTED] wrote: On 9/10/07, Ira [EMAIL PROTECTED] wrote: At 02:11 PM 9/10/2007, you wrote: Can people on this list share their experiences on how they partition a DSL for small business internet service with a router so that a portion is dedicated to VOIP and another portion to computers. Of course, the idea is to do this with a low cost router (under $100). dd-wrt or Sveasoft on a Linksys router though I understand there are better choices in routers today. Don't expect too much out of traffic shaping. While it should work nearly perfectly upstream, there's only so much you can do to control the downstream (from your ISP to you). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which cause less CPU usage: GSM or wav??
Also your Disk subsystem speed. having disk RAM , makes sense in your case. On 9/10/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Barton Fisher wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? G.711ulaw and G.711alaw are the audio transmission methods used for ISDN. If you have a T1 line then the transmission method is G.711ulaw. I've been told that if you play a ulaw signal down an alaw line (T1 signal down E1) then at the other end the voice sounds a bit like a dalek. (Iit's very hard to do this with asterisk since it automatically transcodes between endpoints). The lack of a performance hit is quite striking when you have a recording playing back as a native format rather than being transcoded. (well, it's quite striking when you have thousands of them running simultaneously). Bart Steve Totaro wrote: Michiel van Baak wrote: On 10:28, Sun 09 Sep 07, Barton Fisher wrote: I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram depends on what codec the T1 is using. Best to transcode the ivr sounds to the same codec to prevent on-the-fly transcoding by asterisk. The answer is going to ulaw or alaw depending where you live. T1 should most likely be using ulaw so make everything ulaw, end to end. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2516 (20070909) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broken UDP streams
Maximum retries exceeded on transmission usually comes from NAT issues. you can try this system without NAT and see if problem has resolved. On 9/7/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi All, I'm working from home today (DSL - Internet - 2MB leased line - A*K server behind NAT), and trying to pickup voicemail using Zoiper.. I can access the VM system, I hear all the prompts, and I can even hear part of the message playback. But then I get silence on the call (call stays up), and I get: Parsing '/var/spool/asterisk/voicemail/default/2027/Old/msg.txt': Found -- Playing '/var/spool/asterisk/voicemail/default/2027/Old/msg' (language 'en') Sep 7 13:51:30 WARNING[30737]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. for seqno 2 (Critical Response) Sep 7 13:51:30 WARNING[30737]: chan_sip.c:1245 retrans_pkt: Hanging up call NmM3YmNhNjk0NzhhMjFlYmU5Yzg1YTBmNThlZDNhYWQ. - no reply to our critical packet. == Spawn extension (from-sip, voicemail, 4) exited non-zero on 'SIP/427-b780fa40' On the A8k log. I'm guessing packets are getting lost, but don't understand why it would only be in VM playback that it happens. Any ideas? Adrian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk voicemail to email and relaying
Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows depending on what target mail server requirements are. Base on these facts i came to conclusion of setting up local MTA to relay emails trough another mail server (another mail server beeing their ISP mail server), i dont have very good results with sendmail/procmail and SASL, its inconsitance, works with some provider not all... I was wonderin what do you guys use for your asterisk boxes? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 trunking scalability
Nice to know, luv to have this practical numbers. On 8/28/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Hi, I thought I'd give a follow up to this discussion for the archives... Currently I'm trunking 30 channels of g.729 traffic (no transcoding going on, the call comes in and goes out as g.729) and it takes about 350 kbps bandwith bidirectional. So on average each call takes 11.5 - 12 kbps of bandwith. The solution seems stable and the QoS is identical... so for the price (2 commodity PCs...), IAX2 trunking is well worth the effort since it reduces bandwith usage by a factor of 2. Cheers, Jean-Michel. -- Jean-Michel Hiver - YKOZ +262 (0)692 828 070 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG
Actually i'm using Polycom 501's behind nat and i have no issues. what i usually do is putting static routeable IP for asterisk and using nat and qualify in sip.conf. no issues for me so far. i'm a big fan of Polycom phones, quality of voice, working great with asterisk and low failure rate. On 8/30/07, Dovid B [EMAIL PROTECTED] wrote: - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 22, 2007 4:08 PM Subject: Re: [asterisk-users] Polycom behind NAT won't register to * server behind ALG Henry L.Coleman wrote: I think what Alex was trying to say was that Polycom IP Phones are an example of immature product development. While they look very nice and have a nice display the product doesn't compete very well compared to other manufacturers. The two most obvious flaws are that they cannot be NAT'ed so they cannot be used as Off Premise eXtensions phones and the other being that they take so long to configure and re-boot. I have a golden rule with any phone that I plan on installing for a customerIf I can't get it working within 20 minutes then don't use it. I'm afraid Polycom breaks my golden rule. With such a lot of competition in this market they should have sorted this out two years ago. Reboots should not happen very often. This is a non-issue for most people. I've never seen a phone that could not work with NAT with Asterisk. Polycoms work just fine with NAT and Asterisk. The nice thing about Asterisk's NAT support is that the phone does not need to support NAT. Eric, Try using 5 Polycom's at a remote location behind NAT. Let me know when you need a drink ;) . I had a client with such an issue and the fix was a nice Edgemark or Sonicwall firewall that are set up for SIP and NAT issues. I prefer the Sonicwall. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware download
Thank you David! On 8/26/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: http://www.testforme.com/download/ I'll leave the files there for a few days. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Monday, 27 August 2007 4:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom firmware download Hi: Doug wrote: At 13:29 8/25/2007, Al lists wrote: Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. Which version? http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 330_320.html#download http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip 430.html#download It's funny how every time this question gets asked, there's some smart guy (who doesn't use Polycom sets himself) who finds these links. (I'm sincerely thankful for the effort, though.) Only authorized resellers can download the current firmware from those URLs. The only guaranteed way to get the current firmware is to get it from a/your reseller. Posting the firmware packages on a third-party site is a violation of Polycom's EULA. Why do they do this? Because they want to control the sales channel. I don't agree with it, but it's how they operate. If you want a more detailed answer, ask Polycom directly, and I wish you luck. Cheers, -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom firmware download
Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom firmware download
Thats just sad, I got SIP 2.2 from trixbox now, but still we need to have some sort of place at least for ourselves to download this stuff. Looking for boot loader now. On 8/25/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: On 8/25/07, Al lists [EMAIL PROTECTED] wrote: Hi, I'm trying to use Polycom 330 and apparently it needs latest firmware (SIP 2.2.0). I dont have access to polycom site to download and was wondering if any of you guys have it. Thank you! Best idea is to ask your reseller. I am not aware of a community site with Polycom firmwares. If you wish I think I know the person who created http://spc.pifiu.com and if anyone has any Polycom firmwares I could pass them on. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which OS would be fine for asterisk
What Digium is using is rpath, RHEL /Centos On 8/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Matt Riddell wrote: Steve Totaro wrote: I am bringing up several Fedora Core 7 boxen into production now. Besides a knee jerk reaction that Fedora Sucks, can someone give a real argument as to why I should or should not use it for production? (besides the several MB of yum updates daily, which to me is a good thing). Besides naming a flavor and saying It is the best, can someone add a few statements as to why, which will obviously have to compare the other flavors. At the end of the day, the problem I see with Fedora is that they do things slightly differently from other OSes in the placement of files etc, which can cause headaches you wouldn't see on others. Exactly. I had some difficulties on Fedora as well (can't remember what kind of problem it was - something about zaptel I think) while it just worked for me on Debian or CentOS. (@Steve: So Fedora sucks and Debian is the best ;-) However, there are so many people using Fedora/CentOS/Redhat Enterprise that a quick search of Google will normally reveal the result. While I'm curious if there is a best OS for Asterisk it probably boils down to the simple rule: Use whatever OS you are familiar with and stick to it. If you're used to Debian then CentOS is a bit different too. Unless someone can prove whatever OS is best for Asterisk I'd recommend to use a mainstream distribution. Although I have compiled Asterisk on MacOSX myself this wouldn't be my first choice for a production server - mainly because the whole file system layout is so different and there isn't really an integrated package management. A lot of the differences between distros comes from their choice of package management systems. Once you've used urpmi, yum, up2date, apt-get etc a few times it doesn't really make too much difference which one you're using. Right. But once you need a more complex set of software tools it's a great timesaver to know what the packages are called on a system and what's in there. A word on SuSE: To my impression YaST is an essential part of it. On the one hand I like it but on the other - well, you can shoot yourself in the foot. It tries to be smart and parse all kinds of /etc/* files and doesn't always do a good job. Setting up a DHCP server with some classes and pools for example is almost a piece of cake on Debian. On SuSE it's more like this: Um, I could edit /etc/dhcpd.conf directly but then the next time someone edits the settings with YaST they'd really mess things up - without even knowing. I'm so glad nobody in this thread has argued for using Windows. ;) (It doesn't even come with an ssh client! You really feel like your hands are tied.) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxtel
Is iaxtel still around? I was not able to go to www.iaxtel.com . did the address changed? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some advice
so you are not talking about vanilla asterisk, there are some other applications involved. Paging by nature is resource intensive, but still not sure what else is going on in your system. On 8/14/07, William McCloskey [EMAIL PROTECTED] wrote: The stability problems we have seem to be related to asterisk crashing the apache install on the box when the PHP scripts are performing functions via asterisk. Don't know exactly how they work it all, but that's the gist of it. Best Regards, William J McCloskey Information Technology Manager 503-827-8141 www.timbercon.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, August 14, 2007 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Some advice William McCloskey wrote: I need a quick bit of advice from the list. We purchased an asterisk based phone system back about 6 months ago and we are using Cisco 7940G phones (I know, not everyone's favorites). We are using the second line on the phones for paging with a auto-answer, now my question is having the system call 20 of these paging extensions, should that be enough load to cause instability in the system? Our vendor is claiming it is causing the problems we are having, and I really find that hard to believe. Can you be more specific about the stability problems? That's a bit vague -- it makes it hard to understand what's really happening. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
Cant help you with storm issue but second problem you have is coming from bad FXO module. Replacing that module should fix it. On 8/8/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. Second problem, the card with only 1 fxo port has gone crazy, its permenantly busy, no matter if i reboot the system, even if the system is off, the line is still busy until i unplug it from the digium card. i have no idea whats making the line always busy, this just happened out of no where. again reloading modules, rebooting or even shutting down the system does not make the line un-busy until its unplugged from the card, big problem since its the only line at the location. I appreciate your help everyone. thank you. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
SLA is not BLF. The only thing you need to configure to have BLF is adding hint priority to your dial plan. On 8/8/07, James Collier [EMAIL PROTECTED] wrote: Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS
Re: [asterisk-users] Help : problem in SLA (Shared Line Apperence
Clarify this, what you are trying to achieve?
To see if handsets are being used or not?
Or to see if any trunk is being used or not and share it?
These are 2 different concepts, first is BLF you can have your asterisk to
provide that information with hint priority, and the second one is SLA.
On 8/8/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
On 8/7/07, raviprakash sunkara [EMAIL PROTECTED]
wrote:
Hello Russell,
Nice To meet U and Good Morning. I got u r mail-Id from
http://www.asterisk.org/node/48325
Recently i started the SLA configuration. But i didn't understand the
Flow of its Functionality
One of the My Client Ask to have do deploySLA feature
He Using the Aastra 55i, when users is busy , Aastra 55i will blink
lamps
in SLA.conf
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
Thanks Regards
Ravi Prakash Sunkara
India
--
Thanks Regards
Ravi Prakash Sunkara
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Re: [asterisk-users] Paging Application - Polycom 601
I'm using Page application with Polycom 501 and 601 and have not seen these issue, i would check firmware on 601 and play with couple different firmware. are you checking if the chanavail before sending the Page? On 8/8/07, Bill Andersen [EMAIL PROTECTED] wrote: Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. 1) Has anyone ever seen this problem? 2) Is there a way from the CLI to show and kill a page? 3) Any suggestions? Thanks Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
what you are reading on Cisco manual DN is a completely different concept that what we are dealing in asterisk. In CME you refer to each number as a DN, that concept does not exist on Asterisk. Although Asterisk support SCCP (Skinny) and H323, but its always easier and better to use SIP or IAX. if you like to have a reception phone with BLF, there are lots of options to choose from. Beside the fact that i don't like quality of Cisco Phones, I usually get better and professional results with Polycom. But again that is my opinion. On 8/6/07, Ryan Amos [EMAIL PROTECTED] wrote: The 7914 only works under SCCP; the SIP firmware does not support it at all (the expansion panel won't even power on fully.) The SCCP channel driver under Asterisk doesn't really support the 7914 very well, currently it will only show onhook/offhook state (though there has been much discussion recently about changing this.) If you want to do this with SIP then you're better off with something like the grandstream mentioned, or just use the Flash Operator Panel (IMO it gives you more flexibility at a much lower cost.) I have personally found receptionist phone functionality handled much better with FOP. I have a 7914 and its functionality (and usefulness) is very limited under Asterisk. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *James R. Stevens *Sent:* Monday, August 06, 2007 10:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Learn some terminalogy before mountingthistask. Thank you for your reply as it is exactly what we would need. Sorry I didn't find it myself. I do have a question about configuration within Asterisk. I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated …' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *James FitzGibbon *Sent:* Monday, August 06, 2007 7:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Learn some terminalogy before mountingthistask. On 8/5/07, *James R. Stevens* [EMAIL PROTECTED] wrote: In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? You're probably looking for something like this: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html I have no experience integrating this specific piece of hardware with Asterisk, but I've done what you're trying to do with the Grandstream equivalent for our front reception: http://www.grandstream.com/gxp2000.html and http://www.grandstream.com/gxp2000ext.html As I understand it, so long as the device can do a SIP SUBSCRIBE for each extension you want to monitor and you configure hints in your Asterisk dialplan for those extensions, it should work. You may need to set 'subscribecontext' (in sip.conf) for the phone that will be watching the extensions unless your hints are in the same context as the phone uses for outbound dialing. Of course, what the device does with the various payloads contained in the SIP NOTIFY messages is going to be different for each phone. On the Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red) distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which makes it somewhat useless for transiently connected user agents like softphones. Hopefully someone with experience will speak up and confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE and NOTIFY. If that doesn't work, you could always go with a software solution, like the Flash Operator Panel. voip-info has a list (look at the Operator section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI -- j. -- This message has been scanned for viruses and dangerous content by *Athens Hyperion Scanner*http://www.athensdistributing.com/, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip issue with one way audio
Nat? On 8/6/07, Jason Walker [EMAIL PROTECTED] wrote: I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. any Ideas? Jason ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection
easiest way of connecting multiple Asterisk boxes are trough IP network. I know Digium cards supports HDLC encapsulation but i'm not sure about framerelay. On 8/4/07, Michael Munger [EMAIL PROTECTED] wrote: What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *MOSBAH ABDELKADER *Sent:* Saturday, August 04, 2007 3:16 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Connecting two Asterisk servers with a framerelay connection Hello, Have i to buy an asterisk card like TDM400P to connect the two asterisk servers with frame relay. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Encryption
Iax channel can be encrypted. Not just the authentication, even rtp data, see: http://www.voip-info.org/wiki/view/IAX+encryption On 8/4/07, Michael Munger [EMAIL PROTECTED] wrote: IAX is not encrypted. What you're seeing in wireshark is likely the authentication method you've chosen. (RSA or MD5) You can encrypt it with a VPN as long as you have a pipe fat enough to deal with the overhead a VPN puts on packets. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, July 25, 2007 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Encryption On 23 Jul 2007, at 15:53, Matthew Brothers wrote: I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it looks like the encryption is being setup normally when trunking is enabled. I just can't tell if the voice frame within the trunked packet is encrypted. Any assistance would be appreciated. I thought that Encryption and Trunking are mutually exclusive in IAX. What does the iax debug in asterisk show? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk or asterisknow
You can use both Asterisk or AsteriskNow to have meetme (conference room) On 7/30/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service.You offer me use asterisk or asterisknow. Regards. -- Be a better Globetrotter. Get better travel answers http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=listsid=396545469from someone who knows. Yahoo! Answers - Check it out. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] global variables and updates
Not sure what you are doing with meetme but,
i Always used AstDB() for this type of needs.
On 7/28/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only.
It contains information that may be confidential. Unless you are the named
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and then destroy it.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julian Lyndon-Smith
Sent: Saturday, July 28, 2007 5:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] global variables and updates
Sorry if this appears twice - I originally sent it nearly 18
hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to make sure I don't put two people
together !)
I was going to use a global variable ${NEXTMEETME}, and add one every
time I redirect.
Is the changing of a global variable atomic ? That is, if I
have two or
more channels being redirected at the same time, and they all execute
exten = _X.,n,Set(NEXTMEETME=${MATH(${NEXTMEETME}+1,i)})
exten = _X.,n,Set(MYMEETME=${NEXTMEETME})
if NEXTMEETME is initially 0, would channel A get MYMEETME as
1, channel
B get 2 and channel C get 3, even if they execute the dialplan at the
same time ?
The changing of variables is not atomic as would hope, but there is a
solution for you. Look the application MacroExclusive. Put your Set to
increment the global variable inside of a macro and call it using this,
and you will get the behavior you desire. One caveat, however, is that
you will want as little logic as possible inside of this macro.
MacroExclusive will block all other calls to this macro until the first
one exits. But this is not an issue if all you are doing is a quick
var++ and then leaving.
That's a very nice feature. A quick Google search on the wiki didn't
turn up any topics. Does it queue subsequent calls or just block them
and then logic in the dialplan must be used against a return value?
---
Warm Regards,
Lee
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