Re: [asterisk-users] Experience with virtual servers?
We also run asterisk in a virtual environment, VMWare specifically, along side of web, database, email and DNS (virtual) servers. As far as I'm concerned, it runs as well as it ever did in a real environment. We are using HP Proliant DL360 G5's (3gz Xeon 5160 dual core processors). In our case, the VM hosts that run asterisk are only running Linux guests, and so we require relatively little memory...only 4gb. We also have a larger VM host, similarly configured, but running windows guests and that one has 18gb. Before settling on VMWare, we tried some of the open source solutions, and those did not work as well for us, but VMWare is TOPS. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Friday, April 20, 2012 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Experience with virtual servers? We run many of our asterisk servers on Hyper-V Clusters with openSuse 12.1. They work great All of our PRI PSTN conversions are done with gateway appliances and the bulk of our traffic comes in SIP trunk from providers. We have 16 switches on virtual and 10 on dedicated. We are add all new asterisk switches as virtual, and removing old physical installs as space is needed in the racks to accommodate new servers to support the virtual deployments. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Brynjolfur Thorvardsson bi...@itanet.numailto:bi...@itanet.nu Sent: Friday, April 20, 2012 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Experience with virtual servers? Hi All Does anybody have experience with running Asterisk on virtual servers? I have been experimenting with two suppliers and I am not altogether happy with sound quality etc. Is it perhaps foolish to try and install a production Asterisk server on a virtual machine? With dedicated servers being comparatively cheap (although still several times more expensive than virtual servers), perhaps that is the way I should be going? I have heard someone mention Asterisk friendly VPS providers, how can you tell if they are or aren't friendly? We currently have our Asterisk server running on a five year old single AMD CPU 32 bit machine with 512Mb and that works fine. Even the cheapest virtual server vendors offer servers that seem much more powerful but after testing I am not so sure any more! Any info would be very welcome! Regards Binni No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 2012.0.1913 / Virus Database: 2411/4947 - Release Date: 04/19/12 -- _ -- 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] Under heavy attack
Me too. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Under heavy attack My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.comhttp://www.ilovetovoip.com www.pbxforall.comhttp://www.pbxforall.com (beta) -- _ -- 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] Should I move to 1.6 or 1.8, or stay with 1.4?
We moved a 1.4 installation to a VMWare environment some time ago and it was fairly uneventful. Still, if it were me, I wouldn't change too many things at once and I would first wait until what I currently run is stable under VM. Once stable, I wouldn't hesitate to upgrade and that's one of the nice things about running in a virtual environment. It's makes upgrades such as that really easy, both from the standpoint of moving forward and reverting back, if necessary. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, August 24, 2010 6:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring and how about their stability and reliability? Or should I stay with 1.4? Sincerely, Zeeshan A Zakaria -- www.ilovetovoip.comhttp://www.ilovetovoip.com -- _ -- 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] Should I move to 1.6 or 1.8, or stay with 1.4?
We now run VMWare ESXi 4.0 on HP Proliant DL360 G5 and have not had any issues. A couple of years ago, we tried OpenVZ, but did not have good results. Don't ask to me explain what the problem was, because that was the problem...we couldn't figure it out. It was just unexplained erratic Asterisk behavior that we did not experience on dedicated hardware. And, we were not using any PRI or other boards...just plain old SIP and IAX. It could have been OpenVZ or it could have been something we did, but the result was the same. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, August 24, 2010 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Did you use VMWare's hypervisor? I have no experience with it but I'll be using Proxmox with no KVM, just OpenVZ because the server's processors don't support hardware virtualization. I have worked for someone before with Asterisk 1.4s running on Proxmox, and there was no issue regarding virtulization of asterisk. Plus I am not using DAHDI or PRI, just plain SIP and IAX. Zeeshan A Zakaria -- www.ilovetovoip.comhttp://www.ilovetovoip.com On 2010-08-24 10:07 AM, Bruce Komito bru...@wpti.netmailto:bru...@wpti.net wrote: We moved a 1.4 installation to a VMWare environment some time ago and it was fairly uneventful. Still, if it were me, I wouldn't change too many things at once and I would first wait until what I currently run is stable under VM. Once stable, I wouldn't hesitate to upgrade and that's one of the nice things about running in a virtual environment. It's makes upgrades such as that really easy, both from the standpoint of moving forward and reverting back, if necessary. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, August 24, 2010 6:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one... -- _ -- 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] Should I move to 1.6 or 1.8, or stay with 1.4?
I'm not saying you will have issues with OpenVZ and Asterisk...just that we did (a couple of years ago) and they went away when we rehosted on VMWare. It may work fine for you. We started out with the free version of VMWare, but soon thereafter upgraded to a licensed version of VMWare Essentials for three hosts. (It's normally a $1000 license, but was on sale at the time for $500). One neat thing we are now able to do is drag and drop a running Asterisk system from one VMWare host to another without rebooting the Asterisk environment. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, August 24, 2010 7:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Thanks for sharing your experience Bruce. I am going to use OpenVZ and hope it'll work fine. Is ESXi free or costs license, just in case OpenVZ won't work. The client I worked for, who was using OpenVZ had pretty moderately busy asterisk servers and didn't have any issues with it. Zeeshan A Zakaria -- www.ilovetovoip.comhttp://www.ilovetovoip.com On 2010-08-24 10:45 AM, Bruce Komito bru...@wpti.netmailto:bru...@wpti.net wrote: We now run VMWare ESXi 4.0 on HP Proliant DL360 G5 and have not had any issues. A couple of years ago, we tried OpenVZ, but did not have good results. Don't ask to me explain what the problem was, because that was the problem...we couldn't figure it out. It was just unexplained erratic Asterisk behavior that we did not experience on dedicated hardware. And, we were not using any PRI or other boards...just plain old SIP and IAX. It could have been OpenVZ or it could have been something we did, but the result was the same. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, August 24, 2010 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Should I move to 1.6 or 1.8, or stay with 1.4? Did you use VMWare's hypervisor? I have no experience with it but I'll be using Proxmox with no... -- _ -- 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] Virtual machine timing (KVM)
FWIW, we recently moved a 1.4.29 Asterisk system onto a VMWare guest machine and with 40+ call legs (20+ calls), it isn't even breaking a sweat. We have had no complaints from users nor have we noticed any degradation in voice quality, be it live, voicemail or conference bridge (with six participants). The underlying hardware is an HP ProLiant DL360 G5 (Xeon 5160 3gz, 2 cores) with 20gb of memory and the VMWare version is ESXi 4. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Friday, February 19, 2010 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Virtual machine timing (KVM) To get MeetMe working properly, I know some sort of timing device provided by the zaptel package is required (even if it means the zt_dummy). But, on a virtual machine I know that the Linux timing won't work as expected. Is it possible to then dedicate a physical device like a USB port or something to the virtual machine to use for the timing interrupts? The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in a Xen environment on CentOS for me, although I haven't been using MeetMe. Have you run into issues with it specifically? Which version of DAHDI are you using? If there are some issues that you have found I would like to know... Thanks, Sean -- _ -- 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] Bandwidth management and ADSL router
As does ZeroShell (www.zeroshell.net/eng). Bruce Komito WPTI Telecom (775) 236-5815 On Tue, 26 May 2009, Michael Graves wrote: m0n0wall and pfsense both do traffic shaping, which forcibly allocates bandwidth for your VoIP traffic. Michael On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote: Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards Bilal ___ -- 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 http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com 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] Echo on SIP to SIP calls?
I know the subject of echo has been discussed ad nauseum, but I think I have a somewhat unusual problem. I am suddenly experiencing occasional echo on SIP to SIP calls. This is a new development and has never happened in all the years we've been running *. The phones involved are not junk phones (Cisco 7960's and Linksys 942's). I don't recall seeing any settings anywhere than have anything to do with echo cancellation on non-ZAP devices. Anyone have a clue where I should start looking? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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] Compiling asterisk-addons-1.6.0 under Debian 2.6.18?
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18 and mysql 5.0? I am unable to get configure to recognize the existance of mysqlclient. Imparticular, when it gets to: checking for mysql_init in -lmysqlclient... it returns no. For the past several releases, I've had to hack or otherwise coerce this to work, but this time, no amount of fiddling with options or hacking the script seems to get me anywhere. And, of course, if configure doesn't think mysql is installed properly, it won't build the addons that depend on mysql. Anyone have this experience and a workaround for it? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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] Compiling asterisk-addons-1.6.0 under Debian 2.6.18?
That was the silver bullet...thanks! Bruce Komito WPTI Telecom (775) 236-5815 On Mon, 23 Feb 2009, Tzafrir Cohen wrote: On Mon, Feb 23, 2009 at 12:26:09PM -0800, Bruce Komito wrote: Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18 and mysql 5.0? I am unable to get configure to recognize the existance of mysqlclient. Imparticular, when it gets to: checking for mysql_init in -lmysqlclient... it returns no. For the past several releases, I've had to hack or otherwise coerce this to work, but this time, no amount of fiddling with options or hacking the script seems to get me anywhere. And, of course, if configure doesn't think mysql is installed properly, it won't build the addons that depend on mysql. Anyone have this experience and a workaround for it? aptitude install libmysqlclient12-dev (or 15, or whatever) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] Sonicwall potentially causing long ping times to SIP phones
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 23 Oct 2008, Bill Michaelson wrote: Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? James Lamanna wrote: Date: Wed, 22 Oct 2008 11:35:12 -0700 From: James Lamanna [EMAIL PROTECTED] Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- 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] Sonicwall potentially causing long ping times to SIP phones
You're absolutely right. I only mention Sonicwall, because those are the ones we see most often and there is a perception out there that, because Sonicwall is the (disputed) leading firewall, it should work. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 23 Oct 2008, Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
[asterisk-users] Occassional garbled voicemail
I recently installed 1.4.21.2 on Debian 2.6.18-6 and since then, I am experiencing occassional garbled voicemail messages. Specifically, what happens is that the first 15-20 seconds of the message is fine, but sometimes after that the sound starts to break up and the end of the message is unintelligable. There doesn't seem to be any pattern to this. It happens with equal frequency on incoming calls from both SIP trunks and PRIs. I am *not* experiencing any sound breakup on live calls, either on- or off-net. Has any else seen anything like this? TIA Bruce Komito WPTI Telecom (775) 236-5815 ___ -- 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] Setting up ring group
Sounds more like a hunt group than a ring group. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 31 Jul 2008, Ruddy G. wrote: Why don't you just call the Dial application for each user, one after another ?? The ones that are busy will just go through. So, on the next priority, you dial another one. Tom Moore wrote: Hi guys, What's the best way to setup a ring group that contains 6 extensions so that when a call comes in there starts a 30 second timer and the first available device is rang instead of ringing all extensions at the same time? What I want it to do is cycle through the extensions and have the system ignore the ones that are busy and if there are not any free extensions in the ring group to have the system drop the caller to voicemail. If none of the extensions are present in the group I'd like to also drop to voicemail. Basically what I'm looking for is a multiple extensions version of the standard extension macro with multiple devices and the exten busy state ignored. Tom ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] PSTN to SIP
If your requirements are simple and you only have a small number if E1s, you can also use a Cisco 36xx with a T1/PRI card. 3600's have limited capacity but we run 4 PRIs on a 3640 no problem and it's been very stable for several years. The nice thing about 3600's is they are almost free, although the cards are not. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 17 Apr 2008, mark morreny wrote: Dear all, A quick question on deploying Asterisk over E1. I am looking for a low-cost solution for bridging my E1 line and Asterisk with reasonable stability suppoing both voice and fax. Will a Digium T100 be good for that or I really need a Cisco AS 5400 for this task? What is the difference between using a Digium card vs a physical gateway server? What other alternatives are available? Your suggestions will be greatly appreciated. Thanks, Mark ___ -- 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] Off topic...AOCN wanted
For those CLECs out there, if you know of a contract AOCN that you have personal experience with and would recommend, please reply. For those who don't know what an AOCN is, please delete this message. Bruce Komito WPTI Telecom (775) 236-5815 ___ --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 MWI's will not turn off
I have seen this with Polycoms, ZIP2s and occassionally with Linksys 941s, but only intermittently. Sometimes a powercycle will clear it and sometimes not. We've never figured out what's going on, but we think it is something to do with NAT and the phones not exactly sticking to the spec, but that's only a presumption. Bruce Komito WPTI Telecom (775) 236-5815 On Wed, 28 Nov 2007, Thermal Wetland wrote: Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog ' [EMAIL PROTECTED]' in 32000 ms (Method: NOTIFY) Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:5060: NOTIFY sip:[EMAIL PROTECTED]:33475 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0c3d9f34;rport From: anonymous sip:[EMAIL PROTECTED];tag=as33238a01 To: sip:[EMAIL PROTECTED]:33475 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 88 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- SIP read from y.y.y.y:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0e58862b;rport From: anonymous sip:[EMAIL PROTECTED];tag=as69473f09 To: sip:sip:[EMAIL PROTECTED]:5060;tag=D888A873-3AA22F98 CSeq: 112 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:33475 Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0130 Content-Length: 0 Everytime the phone re-registers these messages are sent and the phone 'beeps' and will turn the MWI indicators on even if they have been manually turned off. Anyone see the issue or have any suggestions? Thanks, 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] Strange asterisk message Remote host can't match request NOTIFY...
For some time, I've been getting the following messages continuously from one of the LANs that I have a number of phones on: [Nov 10 10:09:51] WARNING[32945]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. Although the message doesn't specifically mention a SIP extension by name, I have determined these messages are coming from a number of ZIP2 phones attempting to subscribe to message waiting service. One reason I'm pretty sure about this is that the MWI light on the ZIP2s doesn't light when it should. The phones are connected behind a Cisco router running NAT. I have other ZIP2 phones on other networks that do not have this problem, and I have other non-ZIP2 phones on this network that also do not have this problem. As a result, I have concluded that it is some combination of factors having to do with the ZIP2 AND Cisco NAT together, but beyond that, I haven't a clue. Has anyone ever seen a message like this, and/or understand the cause and, better yet, the solution? TIA! Bruce Komito WPTI Telecom 775-236-5815 ___ --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] Stuck Voicemails?
We used to have this problem with 1.2, too. I think it was some timing thing that resulted from the caller hanging up at just the right (or should I say, wrong) moment, like after the min-message-len timer. I won't tell you what we did to fix it, because you don't want to hear about upgrading to 1.4! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 29 Oct 2007, Matt wrote: This question is about 1.2.x asterisk. Please no flames, or you should upgrade to 1.4. Does anyone know what might be the cause for 'stuck voicemail's in 1.2.6asterisk? By stuck, I mean the phones show a voicemail, and if you log in you get you have 1 new voicemail, and if you delete it it says 'deleted', however it remains. Going into the mail directory reveals that there is either a msg0001.txt.tmp or a msg0001.txt file, but no associated wav file. It happens very randomly, not often, and so far has eluded me being able to figure out what causes it. Why does this happen? ___ --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] Sharing lines with multiple buttons in Cisco 7960?
Has anyone come up with a way of sharing a single SIP registration with two or more line buttons on the Cisco 79x0? This is possible on a Linksys 94x, but I haven't found the magic parameter on the Cisco (assuming there is one). TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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] Linksys 941/942 configuration guide
Does anyone have this guide and be willing to share it with me? Thank in advance? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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] Meetme delay?
I recently installed 1.4.5 and I've noticed a recurrence of a problem that I thought was solved long ago, namely a very long (2-4 seconds) delay on meetme calls. That means with two people in the conference room, it takes 2-4 seconds for what one person says to reach the other person. Is anyone else having this problem, and if so, is there a fix or solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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.anything on FreeBSD?
I was very pleased to learn that 1.4.5 has been released. Unfortunately, I have been beating my head against a wall trying to install 1.4.4 on FreeBSD (6.2). If you have been successful in building 1.4.anything (including addons and zaptel-bsd-trunk), could you please respond, on- or off-list with the secret? Plain vanilla asterisk is ok (I think), but when I try to build asterisk-addons-1.4.1 and zaptel-bsd-trunk, nothing seems to be in the right place (or at least in the place where the build scripts expect it to be) and the build fails. I've tried moving stuff around, creating symbolic links, you name it, but unwinding the build spaghetti is beyond my capabilities, I'm afraid. TIA [If you don't have any experience with FreeBSD, please don't bother responding!] Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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] Linksys 941/942 reboot and persistent MWI
We've got a bunch of Linksys 941/942s and have them all configured to upgrade the config periodically. Problem is, when the phone loads a new config it goes through what appears to be a soft reboot, although it only takes about 5 seconds. During this time, the display goes blank and the (normally) green line buttons flash off briefly. This is a minor nuisance and elicites questions and complaints from users. But, worse, is that when the phone goes through this recycle, the red MWI light comes on. About 95% of the time, it eventually goes off by itself, but occassionally it takes a power cycle to do it. We are running the latest Linksys firwmare. My question is this. Has anyone else experienced this problem and if so, what have you done about it? I can't believe we're alone, as there must be a bezillion of these phones connected to Asterisk systems. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL cmd % pattern matching
Try prefixing the % with a \.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 4 Dec 2006, Garth van Sittert wrote:
Hi Jon
No luck - it works with the quotes and no % sign but as soon as I add
the % it doesn't work.
Garth
Jon Farmer wrote:
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Garth van Sittert [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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[Asterisk-Users] Wanted: Cd-bootable Fedora+Asterisk
I'm in search someone who would be interested in developing a Fedora-baed Asterisk system that is bootable from a CD or possible flash. I am aware of the various commercial and free solutions out there, but none I have found suit our needs...mainly because they are not easily extensible and/or upgradeable. If you are interested in working on such a project, please contact me off-list. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Broadvoice help??
Try register = 7723821447:[EMAIL PROTECTED]/7723821447 That works for me. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 20 Dec 2005, Shawn Porter wrote: Would someone be so kind as to point out what stupid little mistake I have made. I thought I did everything according to the setup page but I fail to register. HOSTS file contains 147.135.8.128sip.broadvoice.com SIP.CONF [general] context=iaxclients ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet pedantic=no ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to no) disallow=all ; First disallow all codecs allow=ulaw,alaw,g723,speex.ilbc ; Allow codecs in order of preference dtmfmode=inband register = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED]/1001 [1001] ;shawn type=friend host=dynamic ;dtmfmode=inband secret= context=iaxclients callerid=Oghma Consulting 647-283- [666] type=friend host=10.0.0.101 canreinvite=no defaultip=10.0.0.101 context=iaxclients insecure=very [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=7723821447 secret=xxx username=7723821447 insecure=very context=iaxclients authname=7723821447 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no SIP DEBUG Asterisk Ready. *CLI sip debug SIP Debugging Enabled *CLI Dec 20 10:51:51 NOTICE[14126]: chan_sip.c:4017 sip_reregister:-- Re-registration for [EMAIL PROTECTED]@sip.broadvoice.com 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK4168ff8c From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 401 Unauthorized Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK4168ff8c WWW-Authenticate: DIGEST realm=BroadWorks,algorithm=MD5,nonce=1135093911710 Content-Length:0 8 headers, 0 lines Responding to challenge, registration to domain/host name sip.broadvoice.com 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK220b3020 From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=7723821447, realm=BroadWorks, algorithm=MD5, uri=sip:sip.broadvoice.com, nonce=1135093911710, response=2c73b280cd7857c8f6d2b56acd6e71eb, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 401 Unauthorized Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK220b3020 WWW-Authenticate: DIGEST realm=BroadWorks,algorithm=MD5,nonce=1135093911970 Content-Length:0 8 headers, 0 lines Responding to challenge, registration to domain/host name sip.broadvoice.com 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.164:5060;branch=z9hG4bK890c From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=7723821447, realm=BroadWorks, algorithm=MD5, uri=sip:sip.broadvoice.com, nonce=1135093911970, response=fb0d8ac4bc042e67a716976d4f10004f, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 401 Unauthorized Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER From: sip:[EMAIL PROTECTED];tag=as565f9ec4 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK890c WWW-Authenticate: DIGEST realm=BroadWorks,algorithm=MD5,nonce=1135093912150 Content-Length:0 8 headers, 0 lines Dec 20 10:51:52 NOTICE[14126]: chan_sip.c:6854 handle_response: Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as565f9ec4' Destroying call '[EMAIL PROTECTED]' Dec 20 10:52:11 NOTICE
[Asterisk-Users] Cisco 79xx display as busy-lamp field
Has anyone used a Cisco 7940/7960 (with or without a 7914) to display busy extensions and if so, would you mind sharing the XML code to do it? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zultys phones
We've used the ZIP2, 4x4 and 4x5s. The ZIP2s work fine, as do the 4x4s and 4x5s, now that Zultys fixed their firmware to accomodate some things that Asterisk did differently than their own PBXs. Prior to that, the 4x4s and 4x5s would lock up during certain types of transfers (highly reproducable). If you use the latest firmware, you'll be fine. BTW, if you need some ZIP2s, we have about a dozen new units that we ended up buying but not using because the customer upgraded to multi-line phones for some of their users. Bruce Komito WPTI Telecom LLC (775) 236-5815 On Mon, 21 Nov 2005, Roger Hill wrote: Hi All: Has anyone used any of the Zultys SIP phones, the 2x2 or 4x4 perhaps? Roger ___ --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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-11-21%5C6378ee589ce345959994b05dc7ae1bb7C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ --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] llamdas por 4 lineas a eeuu tarifa plana
Yo tambien. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 1 Nov 2005, Carlos Alperin wrote: Si se?or, I AGREE. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Tuesday, November 01, 2005 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana Sergey Okhapkin wrote: AFAIK, the official language of this mailing list is English. Butt out. What's the difference to you if two others want to talk in their native language? English is for sure the language that most of us on the list would prefer to use, but Asterisk is a world-wide kind of thing. If help is being given/gotten, then more power to them, I say. If you can't read it, just delete it and go on! B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-11-01%5C6a5aa4a5f14c43cab4afd0f0740ac125C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF tones from PSTN not reaching SIP device
Greetings, I am PRIs connected to a Cisco 36xx gateway, which in turn connects to Asterisk via SIP. The problem I am having is that DTMF tones originated on the PSTN side are not heard on the SIP device. On the other hand, tones originating on the PSTN side are received by Asterisk when talking to voicemail or an autoattendant. From the Cisco debug, I can see the Cisco sending NTE (RFC2833) RTP packets to Asterisk and it appears that Asterisk is propogating them down to the SIP device. However when tones are pressed on the PSTN side, all that can be heard on the IP phone is silence. I've tried this on three different IP phones (Cisco 79xx, ZIP2 and Sipura) and they all behave the same, leading me to conclude it isn't a phone config problem. Everything (Cisco and phones) are configured for dtmfmode=rfc2833. Anyone got any ideas? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ringing on Cisco 79xx
Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk support of MF trunks?
If you have successfully connected MF trunks from a telco switch, please respond. We are looking to support E911 directly from Asterisk and our 911 trunking to the LEC will be over MF trunks to their 911 tandem. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Caller ID Info from Cisco router to Asterisk
The answer is, YES. We have exactly that configuration using a 3640 running SIP to *. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 8 Aug 2005, Bojan Jeremic wrote: Have you resolved this issue, I have a friend who has a solution that involves using autocreatepeer in sip.conf. I don't like that solution for the obvious security reasons, however I don't know a better one. Please, let me know if you know of a different solution and if you want me to forward you the setup with autocreatepeer. Boyan Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI T1) to Asterisk when using SIP? I've done some reasonable searching of the archives and the wiki. I've found some good examples of Cisco configurations, but most examples relate to FXO ports (and most of the FXO ports are of the variety that do not support caller ID). I was not able to find a definitive answer to this question when using PRI for inbound calls. I'm grateful for any assistance in answering this question. Thanks. -- Tony Kava Senior Network Administrator Pottawattamie County, Iowa This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-08-08%5Ca197b46858b1423b8a9f73a14641ab98C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] ast_config not updating voicemail password
I've been using realtime to store my voicemail configuration in a mysql table for several months now, and have had no problems...until today. A few weeks ago, I upgraded to the latest CVS and today I noticed voicemail is not updating the password when the user changes it through option 0. I'm not sure when this started happening, but I assume it was sometime after I upgraded. Has anyone else seen a problem like this, and if so, what's the solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with astperl primitives say... in astcc
I just upgraded to the latest (as of a week ago) CVS and since them, I've had a problem with astcc. I've traced the problem as far as astcc calling any of the AGI say... routines (say_digits, say_number, etc.). As near as I can tell, the calls are executed, but control never returns to the astcc code that made the call, and as a result, the channel simply hangs (i.e., nothing else happens) and astcc never returns to the dialplan. Has anyone else experienced this or anything like it? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] G729
The Sipura SPA2000 only supports one G729 call at a time. Same with the Linksys PAP2. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 17 Jun 2005, David wrote: Hi All, I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest Firmware) to use G729. In sip.conf I have set disallow=all, allow=g729 If Line1 is in use by an agent, then Line2 won't work and vice versa (Inbound Calls Only). I have 40 license for G729. so there shouldn't be any issue with the license. I'm getting the following error msg: -- Called 2012 -- Got SIP response 488 Not Acceptable Here back from 192.168.10.103 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is 'NOANSWER' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' If I change 2012 to ULAW, it works fine. It seems that I can't have two lines configured as a G729. Do you guys have any idea why this happening? Regards, This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-06-17%5Cd81c0f432a8146fd9b6064a4b2fc65b8C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Rack Mount Server Recommendations
We've tried a lot of different types of boxes, but the best I've found so far has been from SuperMicro. Contact me off list for more specifics. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 19 May 2005, Michael B. Murdock wrote: Is there anywhere (or anyone) who has compiled some recommendations on rack mount servers for Asterisk? We are currently using Dell 2650 and Dell 2850 but are seeing some problems with the raid controllers failing and are now shopping for a suitable alternative. Ideally the server would be 19in rack mount, build with similar quality to the the Dell's, and have a -48VDC power supply option. Oh yeah, and run asterisk like a champ. Any suggestions? -- Mike ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-05-19%5C5733432b6be94cb4816f28f58274cdf5C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Satellite Providers
We looked at this earlier this year and, after evaluating several companies, could not get it to work well enough. The problem didn't seem to be latency, but rather lost packets in the upstream direction. Most of the time, we couldn't even get the phone to register, but even when we could, there was such a large amount of breakup (in the up direction) that it was nearly unusable. We tried low-end, consumer type services and they didn't work at all. Even the high-end services that claim to offer guaranteed bandwidth apparently do not live up to their claims. We tried running G.729, which should only need about 32-40k over a link that claimed to guarantee 64k, and the best we got was broken sound. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 May 2005, Yiannis Costopoulos wrote: Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-05-11%5Cc819e577de1140fbaa62d0a53c83de86C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Satellite Providers
I don't doubt at all what you are saying. We never tested a truly high-end solution such as the one you described, because the cost would have been prohibitive for our application. I'm sure we only evaluated shared solutions. I guess my mistake was believing the CIR claims. At the really low-end, I didn't expect much, since they don't offer ANY CIR. But when they claimed 64k, silly me, I believed it. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 May 2005, Chad Wicker wrote: Well there are several problems in your description of Satellite services. For one you are grouping several differing technilogies together as one. What it seemed like you were testing was a shared bandwidth solution typically used by providers to reduce cost. It isn't uncommon to experience sever delays and packet loss on these types of systems. Alot of these shared providers claim 64k cir then oversubscribe over that. Lies, yes, theift yes, and they get away with it... What you would want to ask for is a SCPC (Single Carrier Per Channel) circuit and you should have much better results, cost? a lot more than these shared solutions. You may want to look into the maritime providers/teleports in the area for this type of service. Delay for a decent circuit should not be over 600 ms and it should be steady. Proof is in the pudding, in a SCPC circuit with a v.35 interface you can run an extended BERT test on it without error. and that's Sync data... I speak confidently on this as we are a provider of VSAT services in the oilfield industry. We are bombarded with these low cost competition and have to defend ourselves daily. Alot of providers sell crap at a decent price. We don't and won't. It hurts our market penetration but we tend to keep customers for a good long time. I can answer a lot of questions on this subject if anyone needs. It's a lot like point to point microwave, they experienced their bandwidth sharing days and they quickly died on the vine. The driving force behind shared solutions is that satellite bandwidth is expensive. Chad C. Wicker Systems Engineer Petrocom [EMAIL PROTECTED] 5/11/2005 1:06:52 PM We looked at this earlier this year and, after evaluating several companies, could not get it to work well enough. The problem didn't seem to be latency, but rather lost packets in the upstream direction. Most of the time, we couldn't even get the phone to register, but even when we could, there was such a large amount of breakup (in the up direction) that it was nearly unusable. We tried low-end, consumer type services and they didn't work at all. Even the high-end services that claim to offer guaranteed bandwidth apparently do not live up to their claims. We tried running G.729, which should only need about 32-40k over a link that claimed to guarantee 64k, and the best we got was broken sound. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 May 2005, Yiannis Costopoulos wrote: Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-05-11%5Cc819e577de1140fbaa62d0a53c83de86C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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 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 This message has been categorized as Indeterminate by Bayesian Analyzer. Please click on this link if this message is a Spam http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-05-11%5C5b4b9ad2019e496995ded0f9813f6c7aC=2 Or on this link if this message
Re: [Asterisk-Users] Packetization
The packet size is a function of the number of milliseconds of sound sent in the RTP packet. I don't know how to force * to change this, but you *can* unilaterally change the RTP packet size on the Sipura. By doing this, RTP packets sent by the Sipura will be larger or smaller than the default (.03 ms is the default), and I know * will swallow whatever the Sipura sends it. So, I know it's possible to change this in at least one direction if you are using a Sipura. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 3 Apr 2005, Matt wrote: IAX is not an option as Sipura devices do not support AIX. Yes, the sipura will handle the different packet sizes... Is it possible to reprogram asteris to do this? On Apr 3, 2005 1:55 AM, Steven Critchfield [EMAIL PROTECTED] wrote: On Sat, 2005-04-02 at 21:16 -0500, Matt wrote: I'm aware that asterisk only supports 20ms packetization rates. Due to the fact that I will be using some voip devices on a wireless network which is highly sensative to framerate.. is there any way I can hard code the packetization rate at say 30 or 40ms and then compile astrisk? If so, can anyone in the know tell me what variables I need to look at to change? Are you sure your other devices support different packet sizes? Are you sure the added delay in audio delivery can be handled decently and not cause added echo? Have you considered what IAX trunking can do for you? It will reduce frame rate as you add channels since each packet will then hold the frames for each of the consecutive calls. -- Steven Critchfield [EMAIL PROTECTED] This message has been categorized as Indeterminate by Bayesian Analyzer. Please click on this link if this message is a Spam http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=2 Or on this link if this message is a legitimate mail http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189C=1 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Broadvoice alternatives
If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 23 Mar 2005, Chris Ford wrote: You should try Fordvoice http://www.fordvoice.org they are cheaper than broadvoice also. and have the same service. - Original Message - From: Vicky Shrestha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 8:31 AM Subject: [Asterisk-Users] Broadvoice alternatives Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ 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 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-03-23%5Cfcdbdcefe0bd47b985a85fd1f91855feC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] * and DirecWay
If you have any experience using * (or VoIP in general) with DirecWay, please respond privately. I am particularly interested in experiences in Latin America. TIA! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Dealing with bandwidth limitations
We have a number of sip-connected customers whose broadband connections have suddenly become, uh, less than reliable. Actually, there is nothing wrong with the broadband connection, but rather the network backbone in the country they are connected through has become bogged down. Although latency between the sip clients and the * server is only 125ms (ping times), it seems larger packets either take longer or get lost completely, and the resulting latency as reported by * is 500-2000ms. The result is broken up sound at one end of the connection. (The other end is fine, but that's probably because the routing between the * system and the sip clients is asymetrical, so the problem apparently exists in one direction but not both.) The sip clients all use G.729. My question is this. Are there any RTP settings that I could tinker with that would improve the quality, perhaps at the expense of delay, by making better use of the limited bandwidth available. The problem is not so much that the bandwidth is limited, but that it is intermittent and inconsistent. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] ANNOUNCEMENT : NEW CallingCardApplicationforAsterisk
That's your opinion, and I'm sure you have good reason for it. However, in order to be widely accepted, any app must support mysql, simply because many environments run mysql as their choice of database, and are not likely to change. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 26 Jan 2005, Manjit Riat wrote: Once you compare Postgress and MySQL you will never want to go back to MySQL. -Original Message- From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 26, 2005 10:07 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk NICE! I understand that it works against Postgress, any idea what it would take to port it to mysql if anything? robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Areski Sent: Wednesday, January 26, 2005 12:05 PM To: Asterisk-Users Mailing-list Subject: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zero Warn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunk note : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belaïd Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-26%5Ca11a48a7097e4bc2b63750fbfbfc6519C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] ASTCC: potential billing issue and fix
I had the same problem, and it's a database issue, not a code problem. Use the character ^ in front of the pattern in the routes table, and I think you will have better luck. E.g., ^1416... will match only numbers that start with 1416. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 22 Jan 2005, Nabeel Jafferali wrote: Before I start, I just want to say this is not necessarily a problem with ASTCC, but may be a problem the way I have set up ASTCC (and possibly the way others have set it up as well). The issue is that ASTCC tries to match the pattern *anywhere* in the called number, not necessarily only at the beginning. I have set up ASTCC Routes like this: 1800 TollfreeTrunk1 0 0 100 1416 Canada Trunk2 0 0 400 - other NANPA codes - 1 USA Trunk1 0 0 400 011971UAE Trunk3 0 0 3000 - other international codes - Now, for other international codes I have not included all the countries, just the ones that are important for now. I has expected to add others as they became necessary. However, today somebody called 011966... which is not one of the included countries. I guess it instead picked up the 1 pattern and billed the call at 4c per minute. To get around this, I tried to add: 011 Other Trunk3 0 0 1 which should have charged $1 per minute for all other countries and sent them out Trunk3. If I call 011966... it works fine. But, if I call 011416... it picks up the NANPA pattern for Canada defined, instead of the non-NANPA catch-all I have defined. I tried to fix the problem by adding 0112 to 0119 patterns (4-digit, to make a better pattern match) to the routes table, so 011416... would pick up 0114 instead of 1416, but it didn't work. Reordering the mySQL table so these 8 non-NANPA catch-alls appeared at the top of the table (before the 1416 and other NANPA entries) fixed it though. -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeelatjafferali.net ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-22%5Cc2348896a2944f408c615aa0e5208995C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] # Transfers.
Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 20 Jan 2005, Asterisk List wrote: Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-20%5Ce78d2d987a5e46cca50a486612386c7fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] ntp Server and Zultys 4X4
For what it's worth, I'm working with Zultys trying to solve this exact same problem. So far, they've told me to take an ethernet trace, because they claim the DHCP option 42 isn't being sent, but I know this is not the case, because the phone knows the time, just not the time zone. There is a setting in the general section of the config file called timezone, which defaults to -480 (minutes off of GMT), but this setting only seems to control the value that you are prompted with when the phone boots. If I get a solution, I'll let you know. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 17 Jan 2005, Ronald Hartmann wrote: Good Day List, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not booted they can not call helpdesk.. Is there anyway to fix this. Please excuse my ignorance if this is an ntp server option I am unaware of. ron ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-17%5C566bc776c215431faea5578aee92675aC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] ntp Server and Zultys 4X4
That was the hint I needed. Try adding this to your dhcp.conf: option time-offset -480 (-480 is for PST, -420 is mountain, etc.) Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 17 Jan 2005, Ronald Hartmann wrote: I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on this and I think the issue may be related to the setting of the Time Offset 3.4. Time Offset The time offset field specifies the offset of the client's subnet in seconds from Coordinated Universal Time (UTC). The offset is expressed as a two's complement 32-bit integer. A positive offset indicates a location east of the zero meridian and a negative offset indicates a location west of the zero meridian. The code for the time offset option is 2, and its length is 4 octets. Code LenTime Offset +-+-+-+-+-+-+ | 2 | 4 | n1 | n2 | n3 | n4 | +-+-+-+-+-+-+ Once I have time to play with this I will check it out.. any feedback is appreciated. Ron -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, January 17, 2005 9:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ntp Server and Zultys 4X4 For what it's worth, I'm working with Zultys trying to solve this exact same problem. So far, they've told me to take an ethernet trace, because they claim the DHCP option 42 isn't being sent, but I know this is not the case, because the phone knows the time, just not the time zone. There is a setting in the general section of the config file called timezone, which defaults to -480 (minutes off of GMT), but this setting only seems to control the value that you are prompted with when the phone boots. If I get a solution, I'll let you know. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 17 Jan 2005, Ronald Hartmann wrote: Good Day List, I have my asterisk box setup to be an ntp server, and my zultys 4X4 phone is able to get the time, however I must first select the TimeZone Offset and then it will use the time setting from my server. This is a hassle because every time the phone reboots the user must answer this question and as you can imagine End users do not know what to do and since the phone is not booted they can not call helpdesk.. Is there anyway to fix this. Please excuse my ignorance if this is an ntp server option I am unaware of. ron ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005- 01-17%5C566bc776c215431faea5578aee92675aC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-17%5C2bdd513b1d584377b2e2902952b365fdC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Realtime
I've found, when upgrading from earlier releases that do not support realtime (e.g., 1.0.1), you must first make install from the asterisk directory before attempting to build asterisk-addons. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 9 Jan 2005, Serge Schumacher wrote: I downloaded latest * stable complile it successfully but when compiling the asterisk-addons the res_config_mysql.so is missing. I followed the instructions on wiki for Realtime. What did you do wrong ? Thanx, This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-09%5C0f0b30b139014d9db98fb2812a5ed046C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Another Asterisk Certification
I'm sure it took several hours, but, hey, he only has to sell one to get his money back (: Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 22 Dec 2004, Luke Catranis wrote: How much time did you waste on that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor Sent: Sunday, August 22, 2004 10:24 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Another Asterisk Certification Alternate Certification For those of you who can't (or won't) shell-out the $3000+ for the 5 day certification class, here's a quicker way AND IT'S HALF THE MONEY! www.metrotel.net/asterisk.htm Asterisk is a good product. Some people need certification. A mature product needs certified professionals. Asterisk is maturing. Remember the Certified Novell Engineers? There a a lot of people that know everything about Novell who never got the white lab coat. There is a place for cetification. It helps all of us, even those who never become certified. -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ 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 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-22%5C3a0f4f41805f4aa297eb4dbf29c2b394C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] NPA NXX data
That is correct, and the last time I checked, they sell subscriptions for a monthly charge (depending on frequency of updates) or a one-time charge of $750 for a single copy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 17 Dec 2004, Dave DeChellis wrote: Jon Bebeau wrote: HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database with City and State. Actually it's for an Asterisk routing app I'm working on. I see several vendors that want a few bucks to those that want an arm and leg. I expect this is published somewhere by some government agency, but Google hasn't got me to it yet. Jon ___ 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 Good luck - I'm pretty sure Telcordia maintains the LERG and I don't believe it's accessable for free. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-17%5C53e80d1c194848ab8d9fb66318b14651C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Asterisk Realtime IAX - Adding fields
If you have iax.conf on /etc/asterisk, the iax configuration will be loaded from there and not from what is specified in the realtime config. Remove the iax.conf file if you haven't already. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 14 Dec 2004, Jason Goecke wrote: qualify= and mailbox= do not work with the realtime configuration engine. It doesn't matter if you specify them in the database, the thread that handles them will never look at the peers you have defined in the database, only the ones defined in iax.conf. --- Thank you. Will this be a permanent situation, or be resolved in future releases? = Jason Goecke www.goecke.net Ph: +31.707.504.634 Mb: +31.707.504.634 Fx: +31.847.598.006 Alt#s: +1.720.946.6451 (US) /+44.844.986.9270 (UK) [EMAIL PROTECTED] ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-14%5C619af6baeb0a40b6b62494c321c223a8C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Asterisk Realtime IAX - Adding fields
I assume we are both talking about the static realtime settings, right? I can't testify to the behaviour of realtime iax, but I know if you have sip.conf or extensions.conf in the * directory, those values will not be loaded from realtime. To answer your question about how to specify [general] values, you do that the same way as all the other values. The category column in the realtime config table defines the bracketed section name of the xxx.conf file. So, your general settings in the sql table will all have the value general in the column category. If you don't have that column defined, you don't have the sql table set up properly. There is a perl script that takes any .conf file and loads its values into the ast_config table. If you would like me to send you that script, let me know. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 15 Dec 2004, Jason Goecke wrote: Hello, Based on the behavior I have seen, the IAX.conf file is necessary, as it is still picking up the [general] section and registeration commands from there. If I remove that, how does one add the general settings and register commands to the realtime database? All I have removed from the iax.conf are the user/peer/friend definitions. Appears to work based on that (with VoIPJet, FWD, TelIAX, etc), just not with Voicepuslse. Jason --- Bruce Komito [EMAIL PROTECTED] wrote: If you have iax.conf on /etc/asterisk, the iax configuration will be loaded from there and not from what is specified in the realtime config. Remove the iax.conf file if you haven't already. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 This message has been categorized as Indeterminate by Bayesian Analyzer. Please click on this link if this message is a Spam http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-15%5C2db3c0f577cb4919b8550ef3f2b7bd13C=2 Or on this link if this message is a legitimate mail http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-15%5C2db3c0f577cb4919b8550ef3f2b7bd13C=1 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Realtime problem
I'm having exactly the same problem. I have sip.conf rows in the sql table (ast_config), and removed the /etc/asterisk/sip.conf file. Now I have no sip devices. It's as though realtime is not looking for the sip.conf rows in the table. This is my extconfig.conf: [settings] ; Static configuration files: ; file.conf = driver,database[,table] sip.conf = mysql,asteriskcdrdb,ast_config voicemail.conf = mysql,asteriskcdrdb,ast_config This is my res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asteriskcdrdb dbuser = asterisk dbpass = none dbport = 3306 dbsock = =/var/lib/mysql/mysql.sock These are the startup messages I get when I start * (not voicemail.conf is loaded via mysql but not sip.conf: Dec 14 15:31:01 NOTICE[8102]: res_odbc loaded. Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine odbc Dec 14 15:31:01 NOTICE[8102]: Registered Config Engine mysql Dec 14 15:31:01 NOTICE[8102]: Unable to load config sip.conf, SIP disabled Dec 14 15:31:01 WARNING[8102]: Unable to open IAX timing interface: No such device Dec 14 15:31:01 ERROR[8102]: Unable to load config iax.conf Dec 14 15:31:01 WARNING[8102]: Unable to get our IP address, Skinny disabled Dec 14 15:31:01 WARNING[8102]: Unable to open /dev/dsp: No such device Dec 14 15:31:01 WARNING[8102]: Requested contexts didn't get merged Dec 14 15:31:01 NOTICE[8102]: Loading Config voicemail.conf via mysql engine Dec 14 15:31:01 WARNING[8102]: MySQL database sock file not specified. Using default Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 14 Dec 2004, Clay Reiche wrote: I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' And my device(s) won't register. I don't even see them attempt the registration...(from the CLI in ery verbose.) Maybe I'm not using the right version of asterisk??? Is that possible and how would I know? My show version gives me this: *CLI show version Asterisk CVS-v1-0-12/08/04-16:50:05 built by [EMAIL PROTECTED] on a i686 running Linux *CLI Any help would be appreciated. Thanks! Clay Reiche This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-14%5C45f16737f297472c8726ed904c2e44c6C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] MySQL
If you do: cvs checkout asterisk-addons (without the -r v1-0), you'll get everything you need...including res_mysql.conf.sample . Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 13 Dec 2004, Bill wrote: Same here. I've deleted and re-installed asterisk a few times and the RealTime voicemail never works. The best I've gotten is the MySQL query to execute with the wrong context. When I use cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds to download the latest version the res_mysql.conf.sample isn't even there. I made it from scratch but it still doesn't work. If that file isn't there what else is missing? Bill - Original Message - From: Greg - Cirelle Enterprises To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, December 12, 2004 12:50 PM Subject: Re: [Asterisk-Users] MySQL At 06:29 PM 12/9/04, you wrote: Sure. (I really need to write a wiki on this.) You have two choices here before we start. You can use RealTime one of 2 ways: ODBC or direct MySQL. Currently these are the only two supported methods. Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm going to instruct on how to use/install it. The RealTime MySQL driver can be found inside asterisk-addons. Just do the standard make, make install. Now copy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf (or whereever your conf dir is). Edit the res_mysql.conf to your liking. Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime config stuff. If you want voicemail, add this line: voicemail = mysql,asterisk,voicemail_users No such file res_mysql.conf only cdr_mysql_conf.sample Greg ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-13%5C353c2f11a9c84a71aaf2d99328c5429eC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] MySQL, CDR with MySQL
I have the same problem, and I assumed it was because MySQL voicemail support is now accomplished through the realtime facility. But, so far, I haven't had a chance to research it further. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 9 Dec 2004, Bill wrote: I'm preparing to roll out Asterisk for the voicemail portion of my VOIP network. This week I downloaded a fresh version from CVS of Asterisk and installed the following MySQL 4.1.7 RPMs directly from Mysql.orgFor some reason after I enable MySQL for CDR and Voicemail in the cdr_mysql.conf and voicemail.conf I don't get any MySQL functionality at all. It almost seems as though MySQL support isn't even being compiled into Asterisk. I found somewhere that the Z Library was required and that is already installed. Can someone clue me in? MySQL-client-4.1.7-0.i386.rpm MySQL-devel-4.1.7-0.i386.rpm MySQL-server-4.1.7-0.i386.rpm MySQL-shared-4.1.7-0.i386.rpm MySQL-shared-compat-4.1.7-0.i386.rpm Bill ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-09%5C7350c93c751f4f218ea4f17c983c7491C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] astcc needs AGI.pm...where is it?
Greetings, I tried to build astcc, but the Makefile is looking for Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be and where it comes from? I've dragged in everything I can think of from cvs, and * is otherwise running fine. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Did anybody experience problems with BroadvoiceIncoming calls
LA seems to be down. Switch to DCA or MIA and you'll probably be OK. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 1 Dec 2004, Bartosz Wegrzyn - asterisk wrote: Hi, I am having problems with Broadvoice incomming calls. Did anybody who use broadvoice as a provider experienced and problems today? I want to make sure if this is my equipment or the service. Thanks Bart, ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-12-01%5C492e9452f1724ced8689400a96e75086C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] How to rid yourself of Broadvoice
I would like to echo Luki's comments. I, too, have several Broadvoice lines and, for the most part, the quality of service has been excellent. And, while they do not officially support *, the one time I needed support, they were helpful, and not once did they use the excuse that * was not supported. From listening to the chatter on the list, my sense is that most Broadvoice problems are configuration-related (on the * side), and that was also the case with me. However, once properly set up, the problems have been few and far between. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 29 Nov 2004, Luki wrote: After two months of no service, dozens of e-mails and phone calls, and canned we don't support Asterisk responses this finally got the job done. Huh, what problems did you have? I am managing 9 Broadvoice lines on two * boxes; no problems at all. They email supports is non-existent (or very slow), but I have been very satisfied with their phone support -- yes, they do not officially support * but are willing to help if they can, and in most cases they area actually knowledgeable enough to do so. They even provide you with an * sample configuration if you ask. --Luki ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-29%5C990cf357400145fdb9c621c4a386620aC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] SIP Registration failed notices
For some time (since pre 1.0), I've been seeing the following messages fairly regularly from some, but not all, of my SIP devices: Nov 23 06:37:59 NOTICE[2568]: chan_sip.c:7645 handle_request: Registration from 'John Doe sip:[EMAIL PROTECTED]' failed for '200.100.50.25' I have a mix of Sipuras, Grandstreams, ZIPs and Ciscos, but the message seems to come mostly from Sipuras. The message doesn't necessarily result in the registration being lost, but that is not always the case, which leads me to think that I've been ignoring this long enough. I looked in the chan_sip code and all I can glean from reading it is that there was something in the REGISTER request that * didn't like, but I can't tell from the message what it is or why it doesn't like it. Does anyone know what this message means, why it appears and what I should do to get rid of it? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Linksys RT31P2
If anyone finds the generic version of this available (i.e., not locked to Vonage), please advise the list of where. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 22 Nov 2004 [EMAIL PROTECTED] wrote: Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652scid=29 -Ron This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-22%5C543f125be9b24494a8d7fa465e02817cC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Digits entered ARE NOT RECOGNIZED by bank's IVR's
If you are using G711, try setting dtmfmode=inband. We've had a lot of
intermittent problems with * apparently loosing or ignoring DTMF when
using rfc2833. It doesn't usually happen at the beginning of a call, but
rather after a number of tones are sent, such as when picking up several
voicemail messages or having a dialog with an IVR. When we changed to
inband signalling, our problems went away.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 18 Nov 2004, Joseph wrote:
On Thu, 2004-11-18 at 12:44 -0800, Jongsuk Lee wrote:
My guess for problem is your extension configuration file .
You are probably detecting dtmf such as '*#' and asterisk does
something before it sends.
my advice is ]add those specific bank number and by pass dtmf detection
stuff.
One grunt for channel is that those # and * are hard corded into
channel. There got to be different way for doing this.
I did some testing with some help, I shut down asterisk and enter in
SPA3000 Line1 default dial-plan: #9,:xx.:@gw0 this gives me outside
line when I press #9. So the connection goes through SPA3K unit and I
was able to access bank's automated system. So I would assume that * is
not configured correctly. My configuration is very simple.
[globals]
pstn-spa3k=10.0.0.150:5065
[outgoing]
ignorepat = 9
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,tr)
exten = _9.,2,Playback(invalid)
exten = _9.,3,Hangup
--
#Joseph
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Re: [Asterisk-Users] Broadvoice number always busy
I found LAX either unreachable or non-responsive for most of yesterday. I switch to DCA and no more problems. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 16 Nov 2004, TELUX wrote: I have been using LAX and getting a LOT of busy signals, i have taken the patch off and works fine. Seth Remington wrote: On Mon, 2004-11-15 at 15:01, Jerry Geis wrote: I am still getting a Busy message when I call in to my broadvoice number. Is anyone else still getting that or found a fix to it? I can call out all I want no problem. This seemed to start happening after the patch was applied. I've applied the patch on two separate * boxes (work and home) and both incoming and outgoing have been working fine. I'm using proxy.dca.broadvoice.com if that makes any difference to you. Does sip show registry show asterisk as registered with Broadvoice? -Seth ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-16%5C7a0081afe2904d76b5856b7351c0cd8dC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Source for generic linksys phone adapter?
I bought a few of these from PC Connection but then when I tried to order more, they claim the product has been discontinued by the distributor...whatever that means. Does anyone know of a source for these that is still shipping them? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] BroadVoice
Same here... Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 13 Nov 2004, Doug Shubert wrote: yes.. started around 12:00 noon EST I get sip_reg_timeout: Registration for '[EMAIL PROTECTED] Does anyone know if this is related to the channels patch? Doug Gary White (Network Administrator) wrote: Anybody else having Broadvoice registration problems today? ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-13%5Ccf62fbdc4a664e39b123d2ef9ce2d9a4C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] SIP ALERT_INFO for distinctive ring
Could you please explain how this allows one to interogate the ALERT_INFO sent to * by another SIP device or host? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 12 Nov 2004, Brian West wrote: You need ot set _ALERT_INFO and yes it works. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kubat, Philip Sent: Friday, November 12, 2004 11:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP ALERT_INFO for distinctive ring Does anyone have SIP distinctive rings working with SIP providers, inbound? BroadVoice allows for several numbers on a single account, which they delivered with distinctive ring over the primary number. All the calls come in with the sip header from as the primary number. It looks like (via sip debug and ethereal) that the SIP header variable ALERT_INFO is set to a ringer type. (I believe this is part of the RFC) From what I can figure Asterisk supports setting ALERT_INFO for sending calls to SIP devices. My question is can I read it for inbound calls? Other ideas? Thanks, Philip ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-12%5C4ec54fa2c072429fbd109ef84f5b150fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] No busy-tone
The Busy show be at priority 102 (n+101). Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 8 Nov 2004, Eric Wieling wrote: Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-08%5C2cc9c71461074051a6775f6d7cfd9a8aC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Broadvoice with multiple numbers
I'm doing this and it works. You're right, all the calls come into the same context, but your dialplan should match based on the dialed number. If that doesn't help you, I'll send you a config snipet. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 2 Nov 2004, Richard Cook wrote: Hey, Is anyone using Broadvoice with multiple numbers? Was wondering if there's a way to send each number to a different extension. It seems that they both come into the same context. You can't specify the dial plan based on the number, doesn't work. Any ideas? -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320 ext 2010 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-02%5C52fd21a2b2f540728455a031a11a83dfC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] SIP via Wireless Ethernet Bridge and Double NAT
I've tried double NAT and it doesn't work. Your only chance is to run everything behind the Netgear bridged. Even then, if you are using wireless bridges, you will need to make sure the arp entries do not fall out of the bridge(s) due to lack of activity. One way to do this is to make sure they are re-registering often. I'm sure there are other ways to deal with this, but that is what worked for me. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 1 Nov 2004, Paul Rodan wrote: Just moved into a new place and it'll take 2-3 weeks for my SDSL to be installed. Anyways, found an unsecured wireless network going through my new townhouse at 30% strength. Found the owner and they said I could share it for a couple of weeks. They have a Netgear, 108mbs 802.11 b/g. So I took a LinkSys WAP54g and put it in Ethernet bridge mode, it took the signal and converted it to Ethernet for me. I then plugged it into my Belkin 4 Wireless Router w/ 4 port switch. So now I'm redistributing the connection in my townhouse. I plugged a Cisco ATA-186 into the Belkin, but it's having problems registering with the Asterisk server. I figured the double NAT was messing it up. I'm getting less than 1% packet loss to the internet, so the link is strong. Cable Modem -Ethernet- Netgear Wireless Router -802.11- LinkSys WAP54G -Ethernet- Belkin Router -Ethernet- Cisco ATA186. I keep seeing sip registration failed requests on Asterisk. I checked and double checked the passwords, its fine. I believe it's that the device gets the UDP packets through to the Asterisk server fine, with the authentication information or whatever; but when the Asterisk server tries to respond via UDP, it doesn't make it through. So it fails. I tried port forwarding 5060:5061 and 1:2 from the Netgear to the Belkin and then to the Cisco, but no luck. It could be the double NAT, or one of them isn't properly NAT'ing in order for VOIP to work. I believe it could be the Netgear, as I think I've used the Cisco behind the Belkin in the past without a problem. Either that, or maybe UDP doesn't work across wireless links so well. My only other thing to try is to put a 5 port switch between the LinkSys WAP54g and the Belkin and plug my Cisco ATA 186 and my Belkin into it. This way the Cisco ATA 186 is only behind the Netgear NAT, not behind the Belkin NAT. Cable Modem -Ethernet- Netgear Wireless Router -802.11- LinkSys WAP54G -Ethernet- 5 Port Switch -Ethernet- Belkin Router -Ethernet- Cisco ATA186 Anyways, just wanted to see if anybody has tried something as exotic or similar? Anybody had problems with Netgear or Belkin NAT devices? Or Wireless links? This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-01%5Ca0e4f22430024294817f5cd9d8d09e64C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Cisco PRI Gateway Problems
I think you are missing a dial-peer voice xxx pots entry. E.g.: dial-peer voice 200 pots description Match all inbound POTS calls incoming called-number T direct-inward-dial I don't think the PRI will pick up the call unless the called number matches a number in one of the pots dial-peers. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 29 Oct 2004, Peder Angvall wrote: I am trying to get a Cisco PRI gateway to send calls to * and it doesn't appear to be working. It is a 2610 running 12.3 IP+. I've got the config in there, I can see calls come into the Cisco using debugs, but I never see it try to connect to *. When I do debugs, I see the called # as the 10 digit # and I see the calling # as my #, but I never see anything on *. Both devices can ping each other and neither is behind a firewall. If I do a sip show registry on the * box, the router is NOT registered, but I never see any error messages either, so it looks like it isn't even trying to register with *. Anybody have any ideas? Here is the relevant config from the 2610. We are being passed a 10 digit # (I replaced the real #'s with 123456 below). voice service voip signaling forward unconditional sip controller T1 1/0 framing esf linecode b8zs pri-group timeslots 1-24 interface Serial1/0:23 no ip address isdn switch-type primary-ni isdn incoming-voice voice no cdp enable voice-port 1/0:23 ! dial-peer voice 1 voip destination-pattern 123456 session protocol sipv2 session target ipv4:192.168.1.2:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:192.168.1.2 Here is my sip.conf: [general] port=5060 bindaddr=192.168.1.2 disallow=all allow=ulaw [192.168.1.1] context=pstn-incoming type=friend host=192.168.1.1 dtmfmode=rfc2833 disallow=all allow=ulaw [3200] context=local-phones type=friend username=3200 secret=3200 host=dynamic mailbox=3200 ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-29%5C5bc66d662f1440aba60e35c11252071dC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Zultys Zip 2 Setup
It was trial and error for us, too. Here's a config that works for us with *. Zultys is only politely interested in supporting their phones with non-Zultys systems: ROMAVERSION 3.52 IF0DHCP DHCP SERVERIP 216.xxx.xxx.xx SERVERPORT 5060 DOMAINNAME wpti.net SERVERREGISTER YES DIALPLAN 9|7xx|50xx|0xxx|*x.|1xx|xxx TRANSPORT_TYPE UDP LINE1PORT 5060 LINE1AEC YES SIP_MESSAGE_WAITING YES SIP_SEND_PRACK NO SIP_URI_USER_PARAM NO OOBTELEVENTS OOB_RFC2833 TELEVENTPAYLOAD 101 DROPVOICE YES SQUELCHDTMF NO ABCDMODE TRANSITION G711UON YES G711UPACK 20 G711USS NO G711AON YES G711APACK 20 G711ASS NO G729ON YES G729PACK 20 G729SS NO AJB_MAXDELAY 100 FJB_DELAY 40 AUTO_JB_SWITCH NO COUNTRY USA NTPSERVERIP 192.43.244.18 TIMEZONE -420 DST YES RINGTONE 1 LINE1NUMBER 90055522368 LINE1AUTHUSER 9005552368 LINE1AUTHPSWD pw3268 LINE1CALLERID John Public 900-555-2368 Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 23 Oct 2004, Me wrote: I bought one of these phones and I am trying to set it up. So far, I have figured out how to get to the web interface but I can't seem to figure out how to set some of the most important things like the Proxy address etc.. The manual is useless for things like this as well as their website. The only thing these folks seem to give instructions on is how to change the volume etc, but nothing related to actually setting up the phone for use with asterisk or anything else. The Uniden phone was pretty much the same thing, virtually zero docs on how to get started etc.. So far the cheapest phone (the GrandStream) has been the most straight forward to setup. I have already boxed up the Uniden which is ashame since it's a great phone. Thing is I can't use it behind a NAT so it has to go back :( I did email them though and ask them if they had the new firmware ready.. -- Start Your Own ISP! http://www.YourOwnISP.com ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-23%5C7c19079bf3e44d948f0e40a70fab4469C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] meetme latency
For what it's worth, I have the same observation. Meetme used to work great, but sometime in the last few (3-4) months, it seems to have developed significant latency. Our echo test is also normal (way under a second), as are non-meetme calls. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 19 Oct 2004, Bob Knight wrote: I am pretty sure that I had used meetme in the past (many months ago) with great results. Small number of users, mixed connections, IAX2 and SIP. For the past month or so, meetme has been a real pain due to very large latency. I can take 2 phones on the local lan and still get many seconds of latency. This makes it really hard to carry on a conversation. If I try to have folks join in over the net, we end up with 4 to 5 second latency. Is this normal, or do I have a problem. I am running 2.6.8ish kernel with no zap hardware. I am using the 2.6ish ztdummy. zttest looks ok. Echo test and phone calls are great. I think it is only when I get into the pseudo zap driver that I start having problems. Is it time for me to check out app_conference? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-10-19%5Cfdb007959f614e6190803a5c35248faeC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Cisco to * problem
I am trying to connect a Cisco 3640 terminating a PRI to * with SIP. When I call into the PRI, the Cisco answers the call and sends it on to *, however there is no audio. The clue is, the following message out of *: Oct 15 07:50:58 NOTICE[1094289728]: chan_sip.c:2679 process_sdp: Content is 'multipart/mixed;boundary=uniqueBoundary', not 'application/sdp' Looking at the * code, this looks like a mismatch of some sort between * and Cisco, but I have tried every combination of codecs I can think if, and the problem doesn't change. Has anyone seen this message, or have a clue as to what it means? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Proper Syntax
exten = 777,1,VoicemailMain([EMAIL PROTECTED]) Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 26 Sep 2004, Henry Devito wrote: I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers and asks for mailbox, then password. Is there a way for the Voicemail to read what extension they are calling from and just ask for the password? I have a person complaining because they have to enter their mailbox number every time they check their voicemail and the old pbx didn't ask for it. I thought I saw this on a post a while ago, but of course now that I need it I can't find it. Thank you Henry This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-26%5Cca2dfc4a03aa424992d9a0dca8957323C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] German Termination and DIDs
Try www.sipgate.de . They have DID numbers available in 14 cities in Germany. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 25 Sep 2004, Klaus-Peter Junghanns wrote: Hi, if i understand german telco regulations right (even for a german that's not an easy task...) then a provider may not assign a DID to a non-local client. This would mean that a provider in Berlin may not assign a DID to a client in Munich. So, assigning german DIDs to foreign clients would not be legal at all. Yeeehahh, regulations rule! :-) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Sa, 2004-09-25 um 22.32 schrieb Eric Jacksch: Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric [EMAIL PROTECTED] ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-25%5Cf6f0534ca2fc4ddf99b1a9bdad8698bcC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Re: Help with strategy for echo cancellation.
Not true, in my experience. We have no analog lines (i.e., no FXO ports), only PRIs, and we have consistent echo problems. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 23 Sep 2004, David Cook wrote: I'd like a good plan for this too, however this problem seems to exist only with analog FXO interfaces. If you have 12 lines, would it not have been cost effective to go fractional T1 then the box would be cleaner and the problem be averted? Quoting [EMAIL PROTECTED]: I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-23%5Ca2e4810afa4a433fafbcb80b7ed0e93eC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Re: Help with strategy for echo cancellation.
Probably the reason you get echo on the Voicepulse calls is because the propogation delay between the IP phone and where the call becomes analog is much greater than over your FXO lines. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 23 Sep 2004, Shilliday, Jim wrote: All this is consistent with Cisco's analysis -- you can have echo without analog ports IF there's an analog circuit at the other end of the call (and there usually is). We're getting echo on outgoing calls through VoicePulse, not on the FXO's that only carry incoming traffic. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Thursday, September 23, 2004 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Help with strategy for echo cancellation. Not true, in my experience. We have no analog lines (i.e., no FXO ports), only PRIs, and we have consistent echo problems. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 23 Sep 2004, David Cook wrote: I'd like a good plan for this too, however this problem seems to exist only with analog FXO interfaces. If you have 12 lines, would it not have been cost effective to go fractional T1 then the box would be cleaner and the problem be averted? Quoting [EMAIL PROTECTED]: I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-0 9-23%5Ca2e4810afa4a433fafbcb80b7ed0e93eC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-23%5Cb8d381f1c16943eb89522ac0e5b1d304C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] Beyond T1
You can't run E1 on a circuit designed for T1. T1 is 24 x 64k = 1.5mb; E1 is 30 x 64k = 2mb Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 16 Sep 2004, Andrew Thompson wrote: Christopher Jacob wrote: All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Depending on where you using the circuits, you might try an E1. It uses the same total bandwidth as a T1(I think), but splits the channels at 56K instead of 64K, yielding more channels. (And now I can't remember the number.) I haven't heard of direct DS3 connectivity... Just stretching my imagination a little bit, you might be able to plug a DS3 into a H323 box, and then feed the IP-side of the calls to asterisk -- Andrew Thompson http://aktzero.com/ ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-09-16%5C8a09dc96117f472aab522092083ad700C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] mysql version of Directory app
I installed the mysql/voicemail addon, and it works very nicely, thank you very much. However, the Directory app apparently still takes it's list of extensions from the voicemail.conf table. That's not so nice, since it means maintaining the same list in two places. Am I missing something, or is there a version of the Directory app that queries the users table instead of the voicemail.conf file? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Zip2 configuration via tftp?
I would like to configure my Zip2 phones via tftp, however the tokens in the config file are (apparently) not all documented. Specifically, the username/password/callerid fields seem to be only configurable via the web interface. I find this hard to believe, but the documentation and examples that Zultys provides don't help. If you have an example of a tftp-loadable config file for the Zip2 that you would be willing to share, I would sure appreciate it. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Semi-OT: Splitting a PRI into two PRI's?
I could be wrong, but according to the Max documentation, drop insert only works on a channelized T1...not a PRI. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 Aug 2004, Nate Carlson wrote: On Wed, 11 Aug 2004, Martin List-Petersen wrote: Shouldn't it be possible to pipe the channels for the MAX through the Asterisk box ? The whole PRI into Asterisk and a PRI cable from a second port to the MAX. I haven't looked much at data calls from Zap to Zap, but it looked like it was possible. It most likely is possible, but I need to avoid that path for reliability reasons - we have data calls up to the Max 24/7, and I'd be in big trouble if I did something to cause those to go down outside of scheduled periods. It actually looks like the Max may do what I need with something called Drop and Insert functionality - I'm researching that. Apparently, this is a rather common feature, but I'm new to using PRI's for this type of purpose - always been a data guy in the past. :) | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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] Voicepulse problems?
Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Experience with this online seller?
We've only ordered from them once, but so far they have surpassed our experience with other (unnamed) resellers. I placed an order with them for two phones at 4:30pm their time. Within 30 minutes, I had a confirmation invoice and a Fedex tracking number, and the phones went out that night. From other sources, we're about 50%. That means 50% of the time, we get our stuff and the rest of the time the order is either lost or significantly delayed. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm about to order some few phones from this place: www.thevoipconnection.com Do you guys have any experience with this store? Thank you Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCH3+XeDVKqIr3GURAs4EAJ4zHpqfAWj5ZmHkg6g/prg5ljAkBQCeIxE1 JqYQcuraeBkWICAFnNwvP4k= =DuVi -END PGP SIGNATURE- ___ 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] Zultys Zip 4x4
I have * working with a 4x4. The only difference I can see is that you don't have a secret configured. You might try that and see if it makes a difference. BTW, don't even think of putting the 4x4 behind a NAT server. It won't work. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Mike Roberts wrote: Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '000BEA801CA6 sip:[EMAIL PROTECTED]:5060' failed for '204.194.36.138' The telephone LCD says SIP registation rejected. My sip.conf file looks like this for the ZIP 4x4 [2153] type=friend ; either friend (peer+user), peer or user context=sip-phones username=000BEA801CA6 ; usually matches the [section] title callerid=Zultys 2153 host=dynamic ; we have a dynamic IP address ;nat=no; there is not NAT between phone and Asterisk canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone ;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) ;incominglimit=1 ; permit only 1 outgoing call at a time [EMAIL PROTECTED] ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs ; listed with allow= does NOT matter! ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729; Pass-thru only unless g729 license obtained Thanks in advance Mike Roberts ___ 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] Zultys Zip 4x4
Here's my Zultys config. This gets loaded from the tftp server when the phone boots: [GENERAL_INFO] greeting_message=Customer time_fmt=%H:%M date_fmt=%a %d %b %y date_time_order=0 timezone=-420 country=USA language=ENGLISH delmtr=._ clear_settings=2 [NET_CONFIG] use_dhcp=no ip_addr=1.2.3.4 subnet_mask=255.255.255.0 default_gateway=1.2.3.5 primary_dns=2.3.4.5 secondary_dns= host_name=zipphone domain=wpti.net ntp_server_addr=192.43.244.18 tftp_addr_fixed=yes tftp_server_addr=6.7.8.9 tftp_cfg_dir=./ [SIP_CONFIG] phone_sip_port=5060 rtp_start_port=15000 device_id=772368 display_name=Joe User use_proxy=yes register_w_proxy=yes proxy_addr=1.2.3.4 proxy_port=5060 registration_expires=300 auth_password=geheim proxy_password=geheim call_park_extension=700 inb_im_enabled=no session_expires=300 subscription_expires=300 Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Mike Roberts wrote: I tried that, and it still doesn't work. On your Zultys 4x4, what SIP parameters other than these did you configure: Outbound proxy = IP of Asterisk Registrar Server = IP of Asterisk Proxy Password = same password used in sip.conf Thanks, Mike -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Thursday, July 29, 2004 4:06 PM To: Mike Roberts Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zultys Zip 4x4 I have * working with a 4x4. The only difference I can see is that you don't have a secret configured. You might try that and see if it makes a difference. BTW, don't even think of putting the 4x4 behind a NAT server. It won't work. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 29 Jul 2004, Mike Roberts wrote: Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '000BEA801CA6 sip:[EMAIL PROTECTED]:5060' failed for '204.194.36.138' The telephone LCD says SIP registation rejected. My sip.conf file looks like this for the ZIP 4x4 [2153] type=friend ; either friend (peer+user), peer or user context=sip-phones username=000BEA801CA6 ; usually matches the [section] title callerid=Zultys 2153 host=dynamic ; we have a dynamic IP address ;nat=no; there is not NAT between phone and Asterisk canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone ;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone) ;incominglimit=1 ; permit only 1 outgoing call at a time [EMAIL PROTECTED] ; mailbox 1234 in voicemail context default disallow=all ; need to disallow=all before we can use allow= allow=ulaw; Note: In user sections the order of codecs ; listed with allow= does NOT matter! ;allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729; Pass-thru only unless g729 license obtained Thanks in advance Mike Roberts ___ 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] source for zultys zip phones?
If you know of a good *reliable* source for Zip phones, please respond, off-list if you prefer. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Changing Transfer key
Amen! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 28 Jul 2004, AJ Grinnell wrote: Has anyone been able to change the way that asterisk performs transfers? Instead of using the # key, I would like to due something else, such as flash. # is just causing too many problems with transfers and menus when calling out. ___ 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] NAT problems with ZIP 4x4
I'm trying to get a ZIP 4x4 working behind a NAT server, talking to * on a public address. When I use the same sip.conf configuration (and same NAT server) that works for Grandstream and Sipura phones, the 4x4 can register and make calls, calls *to* the 4x4 do not make it to the phone. I can see from the sip trace that the sip packets to the phone are being retried by *, but I don't understand why. I can only assume, since it works for other phones, the problem is in the phone config and not *. Would anyone who has experience getting this to work, be willing to share their wisdom? Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Random Dropped Called
Run zttool and see if you the T1 card is missing interrupts. If so, put the following statement in your rc.local : # unmask interrupts /sbin/hdparm -u1 /dev/hda This will tell the ide driver not to mask interrupts while servicing disk i/o and the missing interrupts on your T1 card will likely go away. If this isn't the problem, zttool might still give you a hint if there are problems on the PRI itself. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 20 Jul 2004, Paul Oster wrote: I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs which may indicate a problem Jul 15 15:32:41 WARNING[11276]: PRI: !! Got reject for frame 30, retransmitting frame 30 now, updating n_r! Jul 15 17:03:20 WARNING[11276]: PRI: !! Got reject for frame 95, but we only have others! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 124 now, updating n_r! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 125 now, updating n_r! Jul 15 17:11:56 WARNING[11276]: PRI: Read on 66 failed: Unknown error 500 Jul 15 23:08:37 WARNING[5126]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30406 (Response) Jul 16 05:39:08 NOTICE[11276]: PRI got event: 8 on span 1 Jul 16 06:25:04 NOTICE[5126]: Request to schedule in the past?!?! Jul 17 14:43:43 WARNING[11276]: Ring requested on channel 1 already in use on span 1. Hanging up owner. This issue has had me baning my head on my desk for weeks, any information that you may have that could clear this up will be much appreciated. --Paul M. Oster [EMAIL PROTECTED] ___ 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] Problem Starting RC1
I had the same problem. Before you make install from the asterisk directory, try removing all the files in /usr/lib/asterisk/modules . That should resolve any potential conflicts from stuff left over from the last build. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 19 Jul 2004, Nathan Martinez wrote: Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service asterisk start'. I get the following errors logged to /var/log/asterisk/messages each time I try to start: Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already registered (or something close enough) Jul 19 17:32:26 WARNING[1076227072]: Already have an application 'ParkedCall' Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, returning -1 Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, 'res_parking.so' has use count 1 Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so failed! Any ideas would be great. Thank you, Nathan ___ 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] Brain-dead Grandstream BT102?
Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] IRQ Misses and Dropped Calls?
Turn off interrupt masking in your IDE driver: /sbin/hdparm -u1 /dev/hda That solved the problem for me. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 14 Jul 2004, Brian D'Arcy wrote: Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never had any dropped calls, or any problems. I'm receiving the following messages Jul 2 09:30:03 NOTICE[19475]: PRI got event: 4 on Primary D-channel of span 1 Jul 2 09:30:03 WARNING[19475]: No D-channels available! Using Primary on channel anyway 24! Jul 2 09:30:20 NOTICE[19475]: PRI got event: 5 on Primary D-channel of span 1 Jul 2 09:30:41 WARNING[19475]: PRI: !! Got a UA, but i'm in state 1 In between the D-Channel error notices/warings, I'll see channels 1-23 goto yellow alarm state, then I'll see them clear. It does not seem to coincide with the ~hourly reset of the b channels. I've looked everywhere for what these errors could mean, but I'm coming up empty handed. Could these errors be related to the IRQ misses I'm seeing? I'm only logging about 8 misses a day total. Zaptel and Zapata configs pasted below... [Zaptel.conf] span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us [Zapata.conf] [channels] context=inbound switchtype=dms100 overlapdial=yes signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes transfer=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group = 1 channel = 1-23 Any tips, tricks or debugging methods anyone could provide would be extremely helpful! I'm running CVS-HEAD 7/2 for libpri, zaptel and asterisk, however the problem has been occurring since we took the system live in mid-June. Thanks in advance to anyone who might can shed some light. Brian D'Arcy ___ 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] sample config file for GS BT101?
If you have an example of a config file for a Grandstream BT101/102, I would appreciate if you would share it with me. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Intermittent SIP 404 Not Found response?
I have several SIP devices (Sipuras) that are working fine with *, except for one annoying little problem. Occassionally, after being registered for some period of time, the Sipura returns a 404 Not Found to (I assume) an INVITE request. Of course, this makes the extension appear busy. When this happens, I check the Sipura and it is thinks it is still registered and I check * and it shows registered. If I reboot the Sipura or restart *, the problem clears. It also clears by itself eventually. Has anyone seen this behaviour and/or know how to cure it? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Outgoing CallerID on PRI problems
Regardless of what you send in callerid, your PRI has a phone number
associated with it that you don't see, but is used for billing. This is
so you cannot spoof the LD company into thinking the call came from
somewhere other than from you. I believe the PRI provider can provision
the PRI to use either this hard-wired callerid , or the one you provide.
It sounds to me like your PRI is provisioned as the former. I would talk
to your PRI provider and see if they agree and are willing to change this.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 29 Jun 2004, McInnis, JP wrote:
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600 regardless of the extension that
makes the outgoing call.
We have tried using the variable SetCallerID(${BYEXTENSION}) but still
get the same results.
Any suggestions?
JP McInnis, Director of Technology
Copiah Lincoln Community College
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RE: [Asterisk-Users] Vonage and Asterisk integration
A minute is a minute, except that Vonage's plans are mostly all you can eat (unlimited) for a fixed price. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 29 Jun 2004, Ken Wiesner wrote: Personally I don't understand why this is a problem for them. A minute is a minute. They could do the same thing VoicePulse does with their VoicePulse Connect service and provide a low cost per minute service and make profit on volume. I do this all day long with fax messaging. Part of the problem with Vonage is they don't let you port the numbers they assign so you're pretty much locked into them unless you're willing to change your number. As for their tech support, I've found them to be very unhelpful and not very well trained. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Tuesday, June 29, 2004 12:12 PM To: Steve Kalcevich Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vonage and Asterisk integration I have a Vonage line and their tech support is ok. I think the problem is, they have a very strict policy on what they will and won't allow, and this policy is designed to prevent exactly what many people (myself included) would love to be able to do: Use a single Vonage line to make outbound LD calls for many people. If Vonage gives you access to the login credentials, there's no way to prevent someone from originating calls from an * server and racking up lots more minutes than is normal for a single user. They are protecting themselves, and I don't blame them. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Tue, 29 Jun 2004, Steve Kalcevich wrote: Jay Milk wrote: I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. I never worked with vonage, is there tech support that bad? -- Regards, Steve Kalcevich, This electronic message contains information from Primus Telecommunications Canada Inc. (PRIMUS) , which may be legally privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or e-mail (to the number or address above) immediately. Any views, opinions or advice expressed in this electronic message are not necessarily the views, opinions or advice of PRIMUS. It is the responsibility of the recipient to ensure that any attachments are virus free and PRIMUS bears no responsibility for any loss or damage arising in any way from the use thereof.The term PRIMUS includes its affiliates. Pour la version en français de ce message, veuillez voir http://www.primustel.ca/fr/legal/cs.htm ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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] forced ring on dial?
I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has returned a busy indication. The whole problem seems to be * is not waiting for the proper call progress signal from the sip gateway before giving the caller a ring indication. Is there any way to control this so that * waits for call progress from the gateway before giving the caller the appropriate indication, i.e., ring or busy tone? I have been told this is a result of setting * to forced ring and this should be turned off, but of course, on * it is probably called something else. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Directory dial by name
Directory only reads the number if the voicemail user has not recorded his name. If the name has been recorded, it plays that, instead. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 21 Jun 2004, Harold Workman wrote: Just a quick question. I setup Directory dial by name, and I read it looks at the Voicemail config to determine who you want to connect to. The thing I dont like is when it finds a match it reads the extension instead of their name. Is there a way to have it read the name in the voicemail config rather than the extension? Thanks, Harold ___ 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] No config file?
I'm having the same problem...nothing changed...just the CVS version. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 20 Jun 2004, Aaron J. Angel wrote: I updated from CVS yesterday and now everytime I start asterisk, I get the following message: config loader has no config file so nevermind. What does this mean? It doesn't seem to hurt anything, just a tad annoying to see everytime I run asterisk. ___ 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
