[asterisk-users] FXS Solutions for modems and other non jitter tolerant devices
In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built into the unit. Unfortunately the FXS ports are just ATA's off of Asterisk, no different than a SPA2012 unit. The 550 is getting long in the tooth and very expensive for a few FXS ports, what are you folks doing when someone has a need? It can be a modem for the power company to read the meter, a postage machine that needs to get more postage, an alarm system,etc. Is the customer buying a POTS line and splitting it the only other way? Thanks, Matt -- _ -- 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] Anyone have a reliable T.38 Solution
Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt -- _ -- 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] Anyone have a reliable T.38 Solution
On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or 1.8? Glad to know that the PAP2T has a solid T.38 implementation! -Matt -- _ -- 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] Asterisk 1.8.4 - Google iCal not working
When i reload asterisk, calendar show calendars does not show this. What I am missing? I really need to get this to work! You are missing that you should take out passwords from config files. Hope your gmail account didn't get hacked. -- _ -- 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] fail2ban + asterisk
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote: Dear this note is only for fresh administrators don't think about asterisk security. Do you know where you go to 'un-ban' an IP if they made some mistake? Using webmin I was not able to find the IP address that was was banned. Thanks, Matt -- _ -- 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] fail2ban + asterisk
On Mon, Mar 7, 2011 at 9:15 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: iptables -L -v will give you the IP address that was banned -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanks Jamie, I will look around to see the steps to clear an IP. Do you know if you can do this through webmin? I know there is an iptables plug-in. -Matt -- _ -- 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] Voice mail forwarding enhancement
Aloha, We have added the ability to dynamically forward or send a voicemail to more than one mailbox. Here is the link - https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18835 There is a diff file and a drop-in replacement for app_voicemail.c. Here are some basic instructions for the drop-in replacement: overwrite app_voicemail.c in the apps folder in the source for asterisk 1.4 with the new one make clean make make install restart asterisk Testing was done against 1.4.39.1. If all goes well we will submit a version for hopeful inclusion in trunk. Please give me any/all feedback. Aloha, Matt -- _ -- 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] Forward voicemail to group of people
Aloha, Is there a way to forward a message to multiple people from within the telephone user interface? Now there is only the ability to forward to an individual. I see there is a way to leave a message for multiple people using the dial plan but that is not available when you are listening to voicemail. Thanks! Matt -- _ -- 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] Best way to connect to a MySQL Database
On Mon, Nov 15, 2010 at 1:04 PM, Matt Darnell mattdarn...@gmail.com wrote: Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt Thank you everyone for the tips! -Matt -- _ -- 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] Best way to connect to a MySQL Database
Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt -- _ -- 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] Music On Hold Help
Steve, Did you use this syntax to convert: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql -Matt On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com wrote: Can you post a link to the original? On Sun, 31 Oct 2010, Matt Darnell wrote: Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav Sounds reasonable to me. Do you have issues with all MOH? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Music On Hold Help
We have a customer that does not care for the default MoH. We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. We down sample it to 16bit, 8KHz, Mono. We have tried with Audacity, CoolEdit Pro, VLC. Does someone have a file they can send me that we can test with, or has any tips? Much appreciated, Matt -- _ -- 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] Music On Hold Help
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.comwrote: On Sun, 31 Oct 2010, Matt Darnell wrote: We have downloaded some royalty free music but it sounds 'fuzzy' when we test it with the system. Can you post a link to the original? Here is the original - http://www.makaicom.com/music/gt_30.wav Here is after we downsample using cool edit - http://www.makaicom.com/music/gt-30-ce.wav Appreciate any help. -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Warren, Setting the call limits was my issue. I am on a test machine and didn't have it set. Thanks for the help! -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. Leif, Isn't callcounter for 1.6 and not for 1.4? -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
Warren, I tried using AddQueueMember to add agents. If they a user is on a call asterisk shows: Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers We are using 1.4.36. What did you use to keep track of the extension state? Didn't see any option for that at http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember Thanks for the help. -Matt On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Thanks, --Warren Selby On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote: We have a queue that agents log into through the dial plan. Extension Sip/101 logs in as Agent/101 We have 'ringinuse = no' in the queues.conf file. The issue is that when Ext 101 is on a 'non queue' call (they placed a call, someone called their DID, etc) they still receive queue calls. Is there a way to stop this from happening? -Matt -- _ -- 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 -- _ -- 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] Queue Agent Getting Additional Calls When on the Phone
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote: What version of asterisk are you using and method are you using to login your agents? I recently had this issue with a 1.4.33 install where the agents logged in with agentcallbacklogin. In the end I had to move them away from chan_agent altogether, using dynamic agents and AddQueueMember, which has a parameter for designating a device to keep track of the state for that member. Seems to be working for now. Here is little more console output: localhost*CLI queue show Sales Saleshas 0 calls (max 10) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/101 (dynamic) (Not in use) has taken no calls yet No Callers localhost*CLI core show channels Channel Location State Application(Data) SIP/101-000b s...@macro-tl-userexten Up VoiceMailMain(101) 1 active channel 1 active call 'core show channels' show SIP/101 is use but 'queue show' does not. -Matt -- _ -- 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] Polycom getting DCHP address from wrong VLAN
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas li...@amplisys.ca wrote: One more thing: Make sure that the port going to your data-DHCP server doesn't have the voice VLAN set on it. I troubleshot an installation for a few hours before thinking of this... Interesting, the DHCP server for the voice and data are coming from the same router. The router connects to the switch via a trunk port. I will set up a dedicated DHCP server on a port with a PVID of 50. Thanks for the tip! -- _ -- 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] Update the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 12:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP CP, What version of Asterisk are you running. We are using 1.4. Seems like the patches are for 1.2. -Matt -- _ -- 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] Update the LCD with the callee's name after dialing
On Wed, Jun 30, 2010 at 4:26 PM, Ryan Wagoner rswago...@gmail.com wrote: On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. CP Until Asterisk 1.8 is released this looks like the easiest way to get remote party id working. I have modified the patch to work with Asterisk 1.6.2.9. I have also attached a patch against FreePBX 2.7 to add the necessary changes to the dialplan. I have verified this works on a Polycom 550. Ryan Ryan, 1.8 is going to be pretty awesome! I know some folks on 1.6.2.9 that will be interested in your patch. I hope it gets stable quick. -Matt -- _ -- 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] Update the LCD with the callee's name after dialing
Thank you Andrew, I will check it out. We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com wrote: Remote Party ID in trunk, it works There are hacks for other versions. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell mattdarn...@gmail.com wrote: Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- 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 -- _ -- 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] Update the LCD with the callee's name after dialing
Is is possible with a Polycom phone to update the LCD with the callee's name after dialing them? When you dial ext 103 now, it says 'To:103'...would be nice if could have 'To:Dan Marino' This is the case even when you have a contact for ext 103. None of the phones I have ever tested do this, Polycom, Linksys, Cisco, Grandstream, Yealink, etc. -Matt -- _ -- 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] Getting presence working in 1.6.2
On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote: To make it more clear and less cryptic, we split out the callcounter functionality in sip.conf, so that you could turn on/off the SIP device state tracking without limiting calls, and encouraged people to use the GROUP() and GROUP_COUNT() functions in the dialplan to enforce call limits. But why 'callcounter', it is frustratingly close 'call-limit' and there is no possible way to use logic to determine what it does. If a change was to be made, why not use 'devicestatetracking=yes'? -matt -- _ -- 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] Use a BLF for monitoring
Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. -Matt -- _ -- 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] Popular Gigabit Phones
Most manufacturers charge in excess of $80 to upgrade from a 10/100 switch to a 10/100/1000 switch built into the phone. The cost might have been in the chipset 5 years ago but I can get a 5 port gigabit switch for $30. What are most folks using for people that need gigabit to the desktop and don't want to run another cable? -Matt -- _ -- 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] Popular Gigabit Phones
On Thu, Jan 21, 2010 at 3:30 PM, Jonathan Thurman jthurma...@gmail.com wrote: On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote: Most manufacturers charge in excess of $80 to upgrade from a 10/100 switch to a 10/100/1000 switch built into the phone. The cost might have been in the chipset 5 years ago but I can get a 5 port gigabit switch for $30. What are most folks using for people that need gigabit to the desktop and don't want to run another cable? For our engineering staff we use Polycom SoundPoint IP 560's. Cubes with two drops for heavy users who have to be dual homed were build without VoIP in mind (or an tech department at all for that matter)... I haven't run iperf through them, so I don't have any performance statistics. No one has complained except for our fiscal department, the phones do come at a premium above the standard phones =). -Jonathan Thanks for the feedback on the 560's. Polycom's are very well built phones. I am surprised they don't offer a phone in the 3 series form factor that has gigabit. People must not be asking for it because the only folks that seem to offer it are Polycom and Cisco (not the Linksys rebrand). -Matt -- _ -- 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] Force Jitter Buffer for SIP to SIP calls
On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland thermalwetl...@gmail.com wrote: We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want to enable the jitter buffer for the end points having the trouble. Reading the docs, it seems that the jitter buffer is only used when the end point is connected to an app like voicemail. -- -Thermal ___ -- 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 This is from voip-info.org - http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf It is in the [general] section # Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4) # Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to no. (Added in Version 1.4) It mentions the 'receiving side' which should be the incoming or upload form the clients. As I am sure you saw, it is not mentioned in the peers and clients section. Perhaps setting jbforce to no and jbimpl to adaptive. I am sure you read all that, anyone have any real world experience? Aloha, Matt ___ -- 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] What happened to netxusa?
Anyone know what happened to netxusa? Seemed like they dropped off the web overnight. -Matt ___ -- 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] What happened to netxusa?
On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell astma...@gmail.com wrote: They had a nice booth at Astricon and everything. Haven't heard anything about them going down, this might just be an unfortunate IT management incident. Both their toll free and fax numbers go to a re-order message...seems like the worst. -Matt ___ -- 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 IP321?
A client of mine asked about a Polycom IP321..anyone else heard about it? -Matt ___ -- 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 Productivity Suite
I wish Polycom would hire someone with ergonomics skills. The whole menu system is the most painful ever designed outside entry-level phones. Polycom is an acknowledged leader in sound quality and robust hardware but their idea of a menu sucks rocks and always has. Most of their menus require multiple click just to *exit* without doing anything. The 'x' (delete) button would do nicely with no additional cost. I agree, the menu system is not very intuitive. That always seemed strange because their use of soft keys is excellent during an active call. -Matt ___ -- 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 Productivity Suite
Yes with EFK in the latest firmwares you are able to change the on screen button layout. I used it to bring a Do Not Disturb button to the main screen of the SoundPoint IP330's. I may just be dense but paired with the Administrator and Developer guides from Polycom it was still rather frustrating getting the EFK working. If needed I could post that portion of sip.cfg to get you started. -- Robin D. Rodriguez Systems Engineer Ifbyphone, Inc. Phone: (866) 250-1663 Fax: (847) 676-6553 rrodrig...@ifbyphone.com http://www.ifbyphone.com Robin, That would be great if you could send an example. My email is mdarn...@gmail.com Thanks! Matt ___ -- 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 Productivity Suite
Has anyone been able to do the following: 1. Set the phone to automatically record all calls to the USB stick, now you have to press three keys. 2. Put Record on the main screen when a call is active. This would eliminate having to press the 'more' softkey. Thanks, Matt ___ -- 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 licenses
So, in short, if all my calls were from outside to a G729 enabled phone and vice versa, I would reach the limit at 30/30, NOT 15/15. If you had 30 licenses, yes the limit would be when you needed either 30 decoders or 30 encoders. i.e. 1/30 would max you out. -M+ ___ -- 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] Two way bandwidth test
On Wed, Jul 16, 2008 at 3:07 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Tue, 15 Jul 2008, Matt Darnell wrote: Does anyone know of a bandwidth test that tests the upload with the download? All of the ones I can find will test the upload then the download. I from experience I have found that a 3M/768K DSL can only do about 256K/256K simultaneously. You have a sucky ISP or router. The only way I have of testing it is with FTP uploads and downloads or P2P sharing. I would like something more formal that would keep the upload speed the same as the download. VoIP as you know is symmetric. The one VoIP test I find doesn't tell you how many calls you can handle, just if it is VoIP ready. iperf You run a server on one site, and a client on the other. So on site a: iperf -s -u then on the other site: iperf -c ip.of.site.a -u -b 80K -l 160 That's a one-way test from site B to site A. To do a test both ways, one at a time: iperf -c ip.of.site.a -u -b 80K -l 160 -r To test both ways at the same time: iperf -c ip.of.site.a -u -b 80K -l 160 -d The -b parameter is the bandwidth to use, so start at 80K (one SIP link) and go up from there. The -l is the packet length - VoIP packets are typically 160 bytes. The one thing it can't do it send the packets in a timed manner - simulating an RTP stream... ie. it needs a packets per second parameter rather than a bandwidth parameter, but this is usually good enough to find gross problems with links, I've found. Gordon iperf it is! Thanks for the tip. -Matt ___ -- 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] Two way bandwidth test
Does anyone know of a bandwidth test that tests the upload with the download? All of the ones I can find will test the upload then the download. I from experience I have found that a 3M/768K DSL can only do about 256K/256K simultaneously. The only way I have of testing it is with FTP uploads and downloads or P2P sharing. I would like something more formal that would keep the upload speed the same as the download. VoIP as you know is symmetric. The one VoIP test I find doesn't tell you how many calls you can handle, just if it is VoIP ready. -Matt ___ -- 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] Anyone have pricing on the Color Polycom Phone?
IP670 was just released...about 30% more than the IP650. http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html -Matt On Tue, Apr 29, 2008 at 1:02 AM, Patrick [EMAIL PROTECTED] wrote: On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote: Anyone seen anything on the IP670 the Color Expansion? Great timing. Yesterday I was looking at the IP650 and wondered when the successor to the IP650 would arrive. Do you have a link or more info about the IP670? Thanks, Patrick ___ -- 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] Anyone have pricing on the Color Polycom Phone?
On Tue, Apr 29, 2008 at 1:02 AM, Patrick [EMAIL PROTECTED] wrote: On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote: Anyone seen anything on the IP670 the Color Expansion? Great timing. Yesterday I was looking at the IP650 and wondered when the successor to the IP650 would arrive. Do you have a link or more info about the IP670? Thanks, Patrick No other infojust saw the link on Polycom's site. If you click the link, you get a 404. Will post info if I find it. -Matt ___ -- 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] Anyone have pricing on the Color Polycom Phone?
Anyone seen anything on the IP670 the Color Expansion? -Matt ___ -- 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] Where is the Digium DS3 card?
Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt ___ -- 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 - Buddy Watch not a choice when adding Speed Dial
On Feb 1, 2008 3:53 PM, Thermal Wetland [EMAIL PROTECTED] wrote: Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at Auto Divert.I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add bw1/bw manually to the speed dial file it doesn't change anything. The buttons work well for a speed dial. The icon next the speed dial is 10 dots, in the shape of a keypad. Anyone else experience this? Thanks, Thermal Check your sip.cfg for the line: feature.1.name=presence feature.1.enabled=1 I would imagine that you have enabled=0 -Matt ___ -- 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] To DB or not to DB?
On Nov 28, 2007 8:48 AM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Pros: 1. No need to reload Asterisk when you change settings Is reloading the text based config that dangerous? Is there a memory leak or something? How many times can you reload before you should restart Asterisk? -Thermal ___ --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] Connect two Asterisk boxes through IVR Menu
Hello,
I have two Asterisk boxes, each in a different office. Extensions are 1xx
2xx in office 1 and 3xx if office 2.
I have setup IAX2 trunks between them as well as the Outbound Routes.
Intra-office dialing works great.
I can figure out how to transfer an incoming SIP call to the other office
using the IVR. Transferring to extensions on the same system works great.
I have tried this command every way I can imagine, even hard coding the
extension:
exten = _3xx,1,dial(IAX2/{$EXTEN})
exten = 300,1,dial(IAX2/301)
Is there something else you need to transfer using an IAX2 trunk from an
IVR? The outbound route has 3xx for the pattern that works for extension
dialing, I thought the IVR would use the same method. My Outbound Routes
are called office1 office2, the trunks are called to-office1
to-office2.
Thank you in advance for you assistance.
Thermal
___
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Re: [Asterisk-Users] Can you disable Forward on a Polycom phone?
Matt,Wouldn't they have to actually enter a forwarded number for the forward to activate? I've hit the forward button myself many times after a call ends, and the phone asks you for a new number to forward to. Douglas.You are correct you have to enter something as the contact and then press enable..The users must panic and just press buttons to make it go away. I really wish the re-map buttons worked, that would be an easy way out - or if the screen had forward active inverted like when you have DND active. If I find something I will let you know.Aloha,Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can you disable Forward on a Polycom phone?
Aloha,Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to.The real issue is the 'end call' button becomes 'forward' when the call endstherefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up pressing 'forward'.bad button layout. Aloha,Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know who is in this picture?
http://www.bethephonecompany.com/documents/itexpo_la/DSC00495.JPGYou need to have been around in telephony for a little while. Aloha,Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone know who is in this picture?
Well that didn't take long!He was a really nice guyI bet it would be a blast to go have a beer with him.We met him at the Internet Telephony Expo.On 11/2/05, Dean Collins [EMAIL PROTECTED] wrote: Captain Crunch J http://www.webcrunchers.com/crunch/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Matt Darnell Sent: Wednesday, November 02, 2005 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone know who is in this picture? http://www.bethephonecompany.com/documents/itexpo_la/DSC00495.JPG You need to have been around in telephony for a little while. Aloha, Matt ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with calling Perl AGI interface
I'll second that. Make sure your script is in /var/lib/asterisk/agi-bin and you have the right permissions on it. I really just wanted to reply to your post though to congraduate you, Dan Marino, on your recent induction into the Pro Football Hall of Fame ;) Sorry, wrong Dan Marino! -Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with calling Perl AGI interface
On 8/10/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote: Dan Marino wrote: I have installed the Perl library from http://asterisk.gnuinter.net/asterisk-perl and am wondering how I reference agi-test.agi from extensions.conf I have added exten = s,1,AGI,agi-test.agi but that doesn't seem to do it. Is there a certain directory .agi files should be, is that the problem? Depending on your asterisk install, the agi-bin directory can be somewhere like /var/lib/asterisk/agi-bin or /usr/share/asterisk/agi-bin locate agi-bin is your friend :) Cheers, Jean-Michel. Thanks! I found the agi-bin it is working -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Voip-info.org mirror/translation
http://sites.gizoogle.com/?url=http://www.voip-info.org -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issues with Polycom 1.5.2
From the Wiki: 'There are aready a few bugs in 1.5.2 but more fixes some good new features' Anyone know where the bugs are being listed? I am working through a few issues: 1. When rebooting, the phone will pause for exactly 180 seconds with the screen reading 'updating initial configuration'. I know it is 180 seconds because when I tail the FTP server log the entries are always 180 second apart. I know it can read/write to the FTP server, it updates the bootlog etc. Not sure where that timeout is set. 2. If I try to explicity mention files in 'CONFIG_FILES = .cfg, sip.cfg' I get an error 0x1 or 0x4000 on the phone. For the 180 second thing. I noticed this recently as well, but this was due to us using ProFTPD as our ftp server. According to the release notes, they have fixed 'issues' with the proftpd server. What I did to get around it was disable my firewall, allow all phones to update to the newest version, then re-enable firewall, and reboot to verify it worked, and it seems to be working fine now. All these responses are great, I will add them to the WIKI -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issues with Polycom 1.5.2
From the Wiki: 'There are aready a few bugs in 1.5.2 but more fixes some good new features' Anyone know where the bugs are being listed? I am working through a few issues: 1. When rebooting, the phone will pause for exactly 180 seconds with the screen reading 'updating initial configuration'. I know it is 180 seconds because when I tail the FTP server log the entries are always 180 second apart. I know it can read/write to the FTP server, it updates the bootlog etc. Not sure where that timeout is set. 2. If I try to explicity mention files in 'CONFIG_FILES = .cfg, sip.cfg' I get an error 0x1 or 0x4000 on the phone. Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the Polycom 301, 501 601?
These phones are mentioned in the Sip 1.5 manuals, anyone know what the differences are? Where are you getting SIP 1.5 from? When I log into the Polycom download area, all I can find is 1.4.1. They must have pulled it back.maybe some issues, like 1.3.0 -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the Polycom 301, 501 601?
These phones are mentioned in the Sip 1.5 manuals, anyone know what the differences are? Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
On 4/19/05, Mike [EMAIL PROTECTED] wrote: . close source and we own the code. You are no better then Microsoft. Speaking of an over reaction -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blank voicemails being sent to users
Aloha, Issue: Someone calls into voicemail and hangs up Asterisk does not get the disconnect signal Asterisk records for 10 seconds then hangs up Problem: Asterisk will send the voicemail to the user The email reads that the message is 10 seconds long The email attachement is only about 300 bytes If you listen to the message through the phone interface, it confuses the people here becasue it goes from the time date stamp to the message options Is there a way to get asterisk to drop these messages. The minimum message time is set to 3 seconds; I guess asterisk thinks the message really is 10 secondseven though it is less than one second after it trims off all the silence. Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys PAP2 Dual Incoming Calls
Im facing a strange problem using a linksys-pap2 (two ports) ATA: I cant have two simultaneous incoming calls when i use g729 codec, if i use g711 (alaw) there is no problem, is this a know issue or am i missing something? The PAP2 only supports one G.729 call at a time. Same as the Sipura 2000. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Account Codes with SIP
Hello, Does anyone know of an * plug in that will prompt a user for an account code when they make a long distance call? I see where you can have a static variable, but I am looking for a lawyer bill back type application. Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Account Codes with SIP
Does anyone know of an * plug in that will prompt a user for an account code when they make a long distance call? Look at the Authenticate command. Do you know if the entered string gets printed out with CDR records? Got this from the Wiki 1. accountcode: What account number to use: account?, (string, 20 characters) 2. src: Caller*ID number (string, 80 characters) 3. dst: Destination extension (string, 80 characters) 4. dcontext: Destination context (string, 80 characters) 5. clid: Caller*ID with text (80 characters) 6. channel: Channel used (80 characters) 7. dstchannel: Destination channel if appropriate (80 characters) 8. lastapp: Last application if appropriate (80 characters) 9. lastdata: Last application data (arguments) (80 characters) 10. start: Start of call (date/time) 11. answer: Anwer of call (date/time) 12. end: End of call (date/time) 13. duration: Total time in system, in seconds (integer), from dial to hangup 14. billsec: Total time call is up, in seconds (integer), from answer to hangup 15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY (on some CDR backends, e.g. ODBC, these may be integers) 16. amaflags: What flags to use: see amaflags: DOCUMENTATION, BILL, IGNORE etc, specified on a per channel basis like accountcode. 17. user field: A user-defined field, maximum 255 characters -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Push VLAN to Polycom via DHCP
On Mon, 28 Mar 2005 14:29:15 -0600, Jerry [EMAIL PROTECTED] wrote: On Mar 27, 2005, at 12:10 AM, Matt Darnell wrote: Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? Chicken or the egg! How can the Polycom reach the proper DHCP server if it is not on the correct VLAN? That's why Ciscos and Polycoms support CDP, so the CDP-capable switch can supply the correct voice VLAN. I 'assumed' the phone would reboot with the new VLAN setting and get a new IP address from the DHCP server on the phone VLAN - there would be two DHCP servers. I can't think of any other way to make it work with DHCP. If it isn't designed to work that way, why would they put the option in the DHCP section. -Matt I had always understood that they only supported VLAN discovery via CDP. But reading the 1.4 admin guide it says this... VLAN ID See 2.2.1.2.2 DHCP Menu on page 7 Special Case: Cisco Discovery Protocol (CDP)a overrides Local FLASH which overrides DHCP VLAN Discovery. a. Can be obtained from a connected Ethernet switch if the switch supports CDP. This seems to imply that DHCP can be used to spec a VLAN. Looks like logic has let us down again. After talking with Polycom it doens't do what it reads. Here is the quote from the manual: VLAN Discovery Disabled - No VLAN discovery via DHCP. Fixed - Use predefined DHCP private option values of 128, 144, 157 and 191. If this is used, the VLAN ID Option field will be ignored. Custom - Use the number specified in the VLAN ID Option field as the DHCP private option value. VLAN ID Option 128 through 254 (Cannot be the same as Boot Server Option) The DHCP private option value (when VLAN Discovery is set to Custom). Default is 129. *** I can't see how that could mean anything else. Maybe I should sleep on it. -M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Push VLAN to Polycom via DHCP
Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? I can push the boot server via option 66 but that is about it. I have set it for 'fixed' and tried many different option numbers with a couple differnet DHCP servers. SIP firmware 1.3.4 or 1.4.1 doesn't make a difference. Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Push VLAN to Polycom via DHCP
Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? Chicken or the egg! How can the Polycom reach the proper DHCP server if it is not on the correct VLAN? That's why Ciscos and Polycoms support CDP, so the CDP-capable switch can supply the correct voice VLAN. I 'assumed' the phone would reboot with the new VLAN setting and get a new IP address from the DHCP server on the phone VLAN - there would be two DHCP servers. I can't think of any other way to make it work with DHCP. If it isn't designed to work that way, why would they put the option in the DHCP section. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GR303 with *
There was some talk last June about some folks trying GR303 with *. Asterisk supports GR-303 access concentrators now; I do not know if the support is in stable, or only in CVS HEAD. Asterisk does not know how to act _as_ an access concentrator, however. Do you have an recomendations for the GR-303 concentrator? I was read that the GR-303 protocol is very similar to ISDN-PRI NFAS. I don't understand what there is to support a GR-303 concentrator, it appears it presents a standard T1 to the end device. Do you have any idea how hard it would be to plug the GR-303 circuit directly into *? Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GR303 with *
Aloha, There was some talk last June about some folks trying GR303 with *. Was anyone succesful? Would love to hear about it. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004 support asterisk?
Simple Answer: No. the i2004 uses the proprietary nortel UNISTIM protocol. Asterisk uses SIP, IAX, SCCP, H.323, but not UNISTIM. Complex answer: It depends on how much you really want it. There has been an open-sourced implementation of a UNITSTIM server done by Cedric Hans. It is located at http://www.mlkj.net/UNISTIM/voi.tar.bz2 (Note: I have not tried it myself yet). With some work it could be modified Yup, it works. I took a copy of voi to our local Nortel distributor's office and showed their engineers how their phones can be used without their call server. Honestly, I'm not sure if chan_unistim makes much sense. a) The phones aren't cheap - over here they cost as much as Cisco 7940's. b) Nortel is already going to SIP. Their latest switches all support SIP. I guess in the end they might produce firmware to upgrade the i2004 to SIP. You were able to complete calls from one phone to another? The installation doesn't look that difficult. It looks like it was a lot of work to reverse engineer it. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone ever get the Polycom Microbrowser XML document?
Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone ever get the Polycom Microbrowser XMLdocument?
Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? I'm using the Polycom micro browser big time... I have the parking, the sip users online, agents on queue, the meetme rooms and the calls joined. I don't have the manual... I call the Polycom and the guys didn't provide me information about browser, just XHTML classical browser. I'm trying to make table on this micro browser and doesn't work :( If U find how please let me know. Tables? I am still working on 'hello world'! :) I can't make anything appear in the screen. I have tried tags like: titlehello world/title bodyhello world/body texthello world/text and on and onanything I can think ofjust produces a blank screen Maybe I am missing a header or something. The Cisco XML was relativly straight forward. Can you please post some examples...please please Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone ever get the Polycom Microbrowser XMLdocument?
I can't make anything appear in the screen. I have tried tags like: titlehello world/title bodyhello world/body texthello world/text and on and onanything I can think ofjust produces a blank screen Maybe I am missing a header or something. The Cisco XML was relativly straight forward. Can you please post some examples...please please Try: html headtitleTest/title/head body Hello World! /body /html That worked like a charm. I have never worked with a browser that was so strict! Thanks again. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ***Solved*** Lost Password to Polycom IP500
I am embarased to say that I changed it from 456. Can't seem to find the paper it was written on! :( Hi Matt, Press and Hold: 4, 6, 8, * until it reboots. Once you press this keys, you get a prompt for the admin password! Once you press the 4,6,8, * you can enter the MAC address as the password. Works like a charm, thanks Polycom Support! -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lost Password to Polycom IP500
Does anyone know how to default the admin password on a Polycom IP500? Phone has SIP load 1.3.1 Thanks, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lost admin password on Polycom IP500?
Does anyone know how to default the admin password on a Polycom IP500? Phone has SIP load 1.3.1 I have physical access to the phone. Thanks, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lost Password to Polycom IP500
On Fri, 10 Dec 2004 01:26:32 -0500, Brent Franks [EMAIL PROTECTED] wrote: I think it is 456 - Brent Does anyone know how to default the admin password on a Polycom IP500? Phone has SIP load 1.3.1 Thanks, Matt Brent, I am embarased to say that I changed it from 456. Can't seem to find the paper it was written on! :( -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lost Password to Polycom IP500
On Fri, 10 Dec 2004 01:36:22 -0500, Brent Franks [EMAIL PROTECTED] wrote: Brent, I am embarased to say that I changed it from 456. Can't seem to find the paper it was written on! :( -Matt Hi Matt, Sorry I read your last message too quickly. There is an admin guide at http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP _2004-06-16.pdf On page 9 it states how to reset back to factory defaults. Press and Hold: 4, 6, 8, * until it reboots. Hope this helps. - brent Once you press this keys, you get a prompt for the admin password! I was fooling around with those jumper on the back of the phoneno help -M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comdial PBX -- can use Asterisk as VM box?
On Tue, 7 Dec 2004 12:58:11 -0500, George Herndon [EMAIL PROTECTED] wrote: On Dec 7, 2004, at 8:48 AM, [EMAIL PROTECTED] wrote: ken , i too have a comdial analog pbx. i'm running a seperate vm system and would like to migrate to asterisk. right now, my comdial hands off calls via serial connections to my vm box. i don't really know what i'm talking about, but i'd like to find a solution whereby i could accept the T1s (2 in my case) to an asterisk server, route calls to vm as necessary and then hand station calls out to my existing PBX. some clients could be converted over to new IP phones or software based phones (customer service, if quality is good enough) and some clients would remain analog. if anyone is doing this (or a similar but proven and technically correct workflow) let me know. Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a KeyVoice application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the hardware. Is there any way to get Asterisk to respond to whatever mechanism it is that the Comdial puts out to the Dialogic? Things I've already tried and discarded: DID: the PBX strips off the DID stuff before it gets to the Asterisk box Caller ID: ibid. So, I'm guessing that there's some, for lack of a better word, protocol that must be standardized to some extent, that allows things like the Comdial PBX to talk to someone else's VM solution. Can Asterisk play ball? Thanks! -Ken Has anyone written a module for something like this? You would need to intercept the inband digits coming from the PBX ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] www.voip-info.org
On Thu, 02 Dec 2004 17:31:44 +1100, David Uzzell [EMAIL PROTECTED] wrote: Has the wiki died or is it just my routing to the wiki from Australia? I have not been able to connect to it for the last hour or more :( David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It is struggling...I contacted the maintainer. -M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Application almost there..Dialplan challenges
Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call is completed. I can not seem to get rid of the need to dial a lead digit. I don't need any other digits - i.e. voicemail, park - we aren't using any * 'features' just as a SIP-FXS gateway. Is it posible so I can create templates to collect the number and send the call to the Cisco when the template is completed 911 411 611 1[2-9]XX-XXX-XXX [2-9]XX- . The users are not likeing to have to dial '9' Looking forward to updateing to 1.0.0 Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser
Will you post to the list? -Matt On Fri, 20 Aug 2004 15:32:35 -0500, John Baker [EMAIL PROTECTED] wrote: Still waiting on Polycom for something. Will make it available as soon as I get it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with upgrading 7960 SCCP to SIP
I had this exact issue. The 7.1 firmware has some issue where it won't upgrade a SCCP image. I had to upgrade to phone to SIP 6.3 then to 7.1 That was the key for me. -Matt On Fri, 20 Aug 2004 17:49:42 -0400, Doug Shubert [EMAIL PROTECTED] wrote: what version of SCCP are you running? Cisco support link for converting SCCP to SIP http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#topic2 Kevin Day wrote: On Aug 20, 2004, at 3:21 PM, Jerimiah Cole wrote: Kevin Day wrote: If I put P003-07-1-00 in OS79XX.TXT, the phone tries to tftp XMLDefault.cnf.xml. I've tried every imaginable loadinformation parameter, and can't get the phone to actually grab the image, it just keeps redownloading the XML file. If I put P0S3-07-1-00 in OS79XX.TXT, the phone wants to download P0S3-07-1-00.sbn, which I don't have. If I rename P003-07-1-00.sbn to that name, it downloads the file but then says File auth error. Anyone been in this situation before, or have any ideas what to do? Yeah. IIRC, you need to create a SIPmacaddr.cnf with the line image_version: P0S3-07-1-00. Jerimiah Tularosa Communications The version I have doesn't even look for a file like that. It tries to download a SEPx.cnf.xml, if that's not there it tries XMLDefault.cnf.xml. If neither are present, and the OS79XX.TXT doesn't match the version the phone already has, it just reboots. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser
That was part of my problem. I can now get the 600 to download XML, I tried using http://phone-xml.berbee.com/menu.xml and the phone displays XML Error (1,0) syntax error. I'm guessing this is because the XML files at that location are formatted for the Cisco phones. Anyone have documentation on how to format the XML for the Polycom? Strange how it is not included in the Docs. Cisco puts out a 4-5 page document on the commands. Anyone found it yet? -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Conference Call?
Is the conference going to be recorded for later playback. They keynotes and conferences would be nice as well. Aloha, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is nufone web site down?
Can anyone get to www.nufone.net? Is their VoIP down? -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is nufone web site down?
Must be my ISP. Thanks. -Matt - Original Message - From: Shaun Ewing [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 12, 2004 9:42 PM Subject: RE: [Asterisk-Users] Is nufone web site down? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell Sent: Sunday, 13 June 2004 5:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is nufone web site down? Can anyone get to www.nufone.net? Is their VoIP down? -Matt I don't know about their VoIP, but their site works for me. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hot keypad on a Cisco 7960
Aloha, Does anyone know how to have a hot keypad on a Cisco 7960? It allows you to dial on-hook without press the SPEAKER button. Very handy once you get used to it! -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unblocking incoming SIP
The in-between fix really wasn't a fix. Chan_sip was modified some time ago (5/24 or so) to require authentication for inbound calls also. To turn this required authentication off, you need to add insecure=very to your peer definition. 5/24 is some time ago? The age we live in! -M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
