[asterisk-users] Asterisk, OCS and Caller-ID
Hello Everyone, We've connected OCS to Asterisk via OpenSips, and the voice functionality is working fine. I was wondering if anyone out there who has implemented a similar system would be willing to share any information on how they have implemented this, specifically with regards to URI numbering on the OCS side. So far, we've done this: 1. Extensions in * are 4-digit, and the Caller-ID is configured with this number. We append the area code and prefix for outbound calling. In order for the Communicator to pop-up with the username, we've had to put the full number with a + so that the normalized number will match the AD phone number. I would assume that we could fiddle with the Caller-Id with macros or something to correct this. 2. We really aren't sure how to handle the URI for OCS, since most people will continue to use both systems. Right now, it's their 10-digit number without the +, but I really don't know what a good solution is here. We add a SIP channel to their Dial() statement so that the desk phone and OC ring at the same time; we don't use Trixbox or anything like that, just a vanilla 1.4 installation 3. Calls from OCS use the URI configured for the account, so it's number only. I would assume there is a way to use LDAP with * to match a number to AD information and populate it that way. Overall, I'm fairly pleased with the setup so far. Call quality is good, people like the ability to dial straight from Outlook and OC, and people can use it on the road as a smartphone, without VPN access required. We have a lot of remote users and daily conference calls, so it's going to save money for that sort of thing as well. Thanks, Kevin ___ -- 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] Choppy Audio in One Direction
Hello everyone, We had one of our PBXs crash due to a hardware failure, and rebuilt it with PBX in a Flash. We are using the current versions of libpri, zaptel and *. It's the same server with replacement hard drives - a Dell 2850 with a TE410 T1 card, single PRI. It was running v1.2 for years with no problems at all. I'll call this PBX B. Our main PBX in our corporate office - PBX A - has also been upgraded to the latest libpri, zaptel and * (not running PIAF, but fc7 and a manual installation), although the same problem occurred with the previous version. We also tried different hardware on PBX B with the same symptoms. What is happening is that occasionally, audio from PBX B to PBX A will become choppy, while the audio from PBX A to PBX B is unaffected. It doesn't seem to matter how many active calls there are, and we've tried SIP and IAX trunks with the same result. The issue will go away for awhile, and then return. It just happened again, and ping times with full-size packets are a consistent 85-90 ms. If you call through the PSTN, then the symptoms disappear. It could be a networking problem, but this didn't start happening until the server crash, and nothing has changed in the networking infrastructure. Cat /proc/interrupts show that there are no shared IRQs. We just set the switch port to force 100Mb, Full-Duplex. The OS on PBX B is CentOS 5.2, kernel 2.6.18-92.1.6.el5. Hyperthreading is turned on, just as it has always been on the server. Can anyone give my any ideas as to what might be going on? Thanks for any help, Kevin ___ -- 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] Big difference in CPU utilization with MeetMe
Julian, Thanks for the information. We'll wait for a new version, then. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Yap Sent: Wednesday, May 07, 2008 5:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big difference in CPU utilization with MeetMe There is a bug in 1.4.19.1 with IAX. That's your issue. On Wed, May 7, 2008 at 12:38 PM, Kevin Ragsdale [EMAIL PROTECTED] wrote: Hello everyone, We are building a new * server based on a Supermicro motherboard with a 2.8 Xeon processor and a TE220B card. We're using the PBX In a Flash distribution. What we've found is that with a 4 user MeetMe conference, the CPU usage is consistently around 16%. This in comparison to our existing PSTN gateway * box running 1.09 (it hosts our conferences and terminates our T1s). With 23 users and processing all PSTN phone calls, CPU usage averaged from 3-8%. This is an older Supermicro, with a 2.4 Xeon processor. In both cases, the connections are via IAX trunks from our main PBX here, and in two remote locations. We use g711 u-law only - no other codecs are used. If we connect the same number of users through a PRI connection directly to the new server, the CPU is 1% or less, so obviously we've pooched something. We saw this same behavior when we split off the users to a 1.4x based PBX, and we thought it was the server hardware in the new machine, which was an older Dell 2650. But now we're not so sure. I know this is kind of vague, but can anyone suggest what might be happening? New Server CentOS 5, Kernel version 2.6.18-53.1.14.el5 Asterisk 1.4.19.1, and the SVN Zaptel drivers for the TE220B problems posted recently 2.8 Xeon, Hyperthreading disabled, 4GB RAM, 3Ware 9550SX RAID Old Server Fedora, Kernel version 2.4.22-1.2199.nptl Asterisk 1.0.9 2.4 Xeon, Hyperthreading off, 1GB RAM Thanks for the help, Kevin ___ -- 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] Big difference in CPU utilization with MeetMe
Hello everyone, We are building a new * server based on a Supermicro motherboard with a 2.8 Xeon processor and a TE220B card. We're using the PBX In a Flash distribution. What we've found is that with a 4 user MeetMe conference, the CPU usage is consistently around 16%. This in comparison to our existing PSTN gateway * box running 1.09 (it hosts our conferences and terminates our T1s). With 23 users and processing all PSTN phone calls, CPU usage averaged from 3-8%. This is an older Supermicro, with a 2.4 Xeon processor. In both cases, the connections are via IAX trunks from our main PBX here, and in two remote locations. We use g711 u-law only - no other codecs are used. If we connect the same number of users through a PRI connection directly to the new server, the CPU is 1% or less, so obviously we've pooched something. We saw this same behavior when we split off the users to a 1.4x based PBX, and we thought it was the server hardware in the new machine, which was an older Dell 2650. But now we're not so sure. I know this is kind of vague, but can anyone suggest what might be happening? New Server CentOS 5, Kernel version 2.6.18-53.1.14.el5 Asterisk 1.4.19.1, and the SVN Zaptel drivers for the TE220B problems posted recently 2.8 Xeon, Hyperthreading disabled, 4GB RAM, 3Ware 9550SX RAID Old Server Fedora, Kernel version 2.4.22-1.2199.nptl Asterisk 1.0.9 2.4 Xeon, Hyperthreading off, 1GB RAM Thanks for the help, Kevin ___ -- 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 branch offices through IPsec tunnel --latency effects?
We have two offices - one in Oklahoma and the other in Vancouver, BC - connected via an OpenVPN connection. We have big pipes at each site (15Mb and 10Mb), and it works great. We average about 70ms latency through the tunnel. We have about 5-6 conference calls per day with up to 20 users, and no one has complained about the audio quality (after we upgraded the Polycom firmware, that is. We had a godawful echo problem with the older firmware). We used to have an IPSec VPN, but for us the OpenVPN solution is much simpler, and has been more reliable. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Tuesday, July 25, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Connecting branch offices through IPsec tunnel --latency effects? Hi: If I connect two offices through an IPsec tunnel, what is the impact on latency, and does it noticeably affect calls? Has anyone out there tried this? What were the effects? Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Echo
Hello, We just experienced a problem that we though might be useful to anyone using Polycom phones. We are installing a new system at one of our remote offices and were experienced a ton of echo on our side while the remote side was on speakerphone. It turns out that the desk surface was causing the echo - when the phone was lifted off the desk, the echo disappeared. We also were able to put a mousepad under the phone to eliminate the echo as well. Since the microphone is on the bottom of the phone, there must be some kind of sound reflection going on. Just a heads up - this is the first time we've seen this happen. Kevin ___ --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] 1.6.3 Polycom Firmware?
Has anyone tried the newest Polycom firmware? The release notes indicate they have added support for a new BLA draft. TIA, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1.6.3 Polycom Firmware?
It's Bridged Line Appearance. It's something our Definity PBX system had - an admin could see whether or not her boss was on the phone, and could pick up the line if he/she was out of the office. The buddy feature of the Polycoms can sort of do this now, with the hint feature of * (you can monitor, but not pick up the call). It's something we get asked about all the time. Supposedly, sipx is going to support it in their 3.x releases. Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor Sent: Tuesday, November 22, 2005 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1.6.3 Polycom Firmware? Kevin Ragsdale wrote: Has anyone tried the newest Polycom firmware? The release notes indicate they have added support for a new BLA draft. New BLA draft? Would you mind explaining what that is. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT ***
Pardon my ignorance, but could someone explain to me what the benefits of this patch? We use 1.0.9, and have our Polycoms showing off-hook status using the buddy lists and speed dial. How does this patch improve upon this? Thank you, Kevin Ragsdale -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Sautter Sent: Thursday, August 25, 2005 11:01 AM To: Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: [Asterisk-Dev] Patch 3644 - subscription states *** IMPORTANT *** Olle E. Johansson wrote: We really need test input of the latest patch in this issue report. And we need them today. If you are interested in device state notification in SIP - stop whatever you are doing and give us feedback NOW! Thank you for your assistance! http://bugs.digium.com/view.php?id=3644 PS. Thanks to Xylome for updating this patch so many times! i can only second olle! this patch has a track record since february and i know from all the emails i received from users and the posting to the digium lists that this is an essential feature! so if you need this functionality give feedback to the bugtracker and it will be gladly in asterisk 1.2. regards frank (aka xylome) This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ --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] Siemens SX66 wi-fi handset released
I've got one, and installed Xten's softphone software. It worked fine with a wired headset, but my bluetooth headset won't work with it - a limitation of the OS, not Xten - so it's not really useful for me until that (if it ever does) gets fixed. Kevin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Monday, April 25, 2005 6:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Siemens SX66 wi-fi handset released http://www.pcmag.com/article2/0,1759,1787787,00.asp I thought this new wifi handset may interest a few of you on the list, anyone here actually seen or even better used on of these? Cheers, Dean ___ 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] The Smallest Asterisk Server Ever?
http://nlug.org/mail/nlug__2003_12/0094.html Kevin -Original Message- From: Panny Malialis [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 2:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I cant wait to see the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlugb2003_12/0094.html I've tried removing the b with no luck Anyone know what the link should be ? Thanks Panny - Original Message - From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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 ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
-Original Message- From: Ken Alker [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:59 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea Does something like this already exist for cheap? If so, is it any good? If so, does it need more features? If not, would you buy something like this? If so, what features have I missed? If so, what is it worth? Daydreaming, as usual. Ken Ken, 3Com makes a 24-port midspan box that sells for around $800. Kevin This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ 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] Power Over Ethernet for *any* ethernet switch (or hub); product idea
-Original Message- From: Martin [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 10:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea snip http://www.goldmark.org/jeff/stupid-disclaimers/ -- Art is anything you can get away with. -- Marshall McLuhan. Martin, We have rules in place that remove it from emails to mailing lists, but I fat-fingered the digium address. Should be fixed now. Apologies, Kevin ___ 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
