Re: [asterisk-users] Fwd: Polycom phone time behind one hour.
Insert your offset into this line: tcpIpApp.sntp.gmtOffset= eg - EST (GMT -5) = -18000 Bryan M. Johns Shelton | Johns 678.248.2637 Office 678.810.0730 Direct 678.303.3424 Fax Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Nov 18, 2008, at 5:46 PM, Doug Smith wrote: Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls timezone information and DST. I put in carriage returns to make it easier to read as it is all one line. Can anyone see anything obvious (I have missed after reviewing many times) with this config that would cause my phones to be behind an hour? I tried changing overrideDHCP=0 to a 1 with no luck. SNTP tcpIpApp.sntp.resyncPeriod=3600 tcpIpApp.sntp.address=207.207.*.* (Address replaced with asterisk to protect our server IP) tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=9 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=4 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ Any help at resolving this would be greatly appreciated. Many of our office workers are annoyed that their times are behind an hour now. Thanks, Doug Smith Alchemy Systems ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Scalability
We have multiple installs that tested-out at nearly concurrent 400 SIP channels on a Dell 2950 with 2Xquad core at 1.6 Ghz, 16 GB of RAM. Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Feb 8, 2008, at 5:09 AM, Femi wrote: Hi, Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls Thanks Femi ___ -- 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] Finding difficulty in installing Asterisk
It appears as though SELinux is preventing you from moving forward. Perform the following to disable SELinux. cd /etc/selinux vi config change enabled to disabled write your changes reboot Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Jan 25, 2008, at 3:44 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Dave, I did make clean and then make. But then when I am giving make install its giving error AVC access denied. I am using Fedora. What may be the problem? Help me.. Thanking you, Preeta Pandey -Original Message- From: [EMAIL PROTECTED] on behalf of Dave Cotton Sent: Fri 1/25/2008 1:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Finding difficulty in installing Asterisk On Friday 25 January 2008 05:25:57 Lyle Giese wrote: You need to do a 'make' before the 'make install'. make install will do all that is necessary to install a program including making any files necessary. -- Dave Cotton ___ -- 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 Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk scale to 500-1000 phones?
Jesse, We have multiple installations of this scale and a few with far more concurrent call paths (250+). In our experience, Asterisk scales nicely to these levels as long as you are realistic about what you expect of the server. For instance, we rarely, if ever, convert signal to TDM. We instead use SIP dial tone from a tier-1 carrier. Also, if you expect any substantial amount of meetme conferences, you might want to consider running those on separate hardware. As the numbers go up, you can peel-apart your switch into functional duties such as two SIP switching servers, two voicemail servers, one conferencing server, etc. Just some ideas. Best of luck to you! Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Dec 27, 2007, at 11:33 AM, Jesse Molina wrote: Anyone have opinions on how well Asterisk scales to 500-1000 stations, in regards to reliability, system performance, as well as ease of management? Assume relatively low call volume; let's say two public network PRIs (47 DS0s). -- # Jesse Molina # The Translational Genomics Research Institute # http://www.tgen.org # Mail = [EMAIL PROTECTED] # Desk = 1.602.343.8459 # Cell = 1.602.323.7608 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk install beta testing/config help
Make certain that selinux, iptables and ip6tables are disabled and off. Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Dec 2, 2007, at 3:18 PM, James Cox wrote: I have asterisk up and running on a fedora system but having trouble accessing system via softphone (ekiga and xlite). Im a newbie to asterisk and was looking for some help walking through this. I imagine 10 - 15 mins would be all needed to make proper config changes needed. Once I get this setup I'd be interested in discussing customizations and scripts so any freelancers or companies welcome since the sooner i get this working the sooner can move to that next stage. thanks in advance! My yahoo IM is jameswcox2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Phone and bitmaps
You aren't including the file extension when referencing the graphic name, are you? If so, that would be the problem. You might also want to try loading the parameters to the fields for the 650 also. Just a couple of ideas. Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Shaun R. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 23, 2007 4:52:26 PM (GMT-0500) America/New_York Subject: [asterisk-users] Polycom Phone and bitmaps I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my sip.conf bitmaps IP_500 . bitmap.IP_500.67.name=mylogo/ /bitmaps indicators ind.idleDisplay.enabled=1 ind.idleDisplay.mode= Animations IP_500 IDLE_DISPLAY ind.anim.IP_500.29.frame.1.bitmap=mylogo ind.anim.IP_500.29.frame.1.duration=0/ /IP_500 /Animations Anybody know where i'm going wrong, watched the ftp logs and i dont see the phone downloading the mylogo.bmp either. Nothing in the -app.log either about it. ~Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk under VMWare
As a pure SIP solution, we have switched as many as 120 call paths through a VM on a lightly populated host. Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: WipeOut [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 23, 2007 1:51:23 PM (GMT-0500) America/New_York Subject: [asterisk-users] Asterisk under VMWare Anyone had any experience with an Asterisk server as a VMWare virtual machine? ___ --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] Wondering why I can't post
Stephen, Thanks for the heads-up on the cab ride from Phoenix to the event. I did not know it was that far. I will be coming in Wednesday morning and I may take the same route you are considering. Anybody coming in Wednesday morning that wants to split fare? Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Stephen Bosch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 17, 2007 1:45:00 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] Wondering why I can't post Matt Riddell wrote: Stephen Bosch wrote: I've been trying to post a specific message for the last four or five days. It's on a specific topic, and I suspect the topic is the reason it is not being published to the list. Which would suggest that some kind of keyword filtering is being done, though I've rephrased the message several different ways without success. I'm sending this message to see if my new posts even make it to the list. If this one does, I'll have my answer. Yes this post is making it. Are you bashing someone/something? Anything in the mail likely to get someone in legal trouble? The answer is no to both questions. Here's what I'm trying to post: Subject: Astricon 2007 -- does anybody need a ride? Hi, folks: Steve Totaro and I are going to be sharing a sedan from Phoenix Sky Harbor airport to the conference hotel for the conference. We're arriving on Tuesday night. The conference hotel is 45 minutes away (assuming good traffic); the taxi fare will be a killer. As an alternative, we'll be booking an executive sedan. We'll have room for one or two more people; if we fill it to the published maximum (4 people), the cost per person will be a very reasonable 19 USD per person, not including taxes and tip. If you'll be arriving on Tuesday evening and are interested, please contact me off-list. Cheers, Stephen Bosch ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overhead paging over IP...
You can use an inexpensive PC with a sound card. Install Asterisk on it and set an extension that calls /dev/dsp. This will send audio out the speaker port on the sound card. Setup a trunk between this unit and your primary Asterisk server and you should be in business. Bryan M. Johns Shelton | Johns Technology Group Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com On Sep 4, 2007, at 7:07 PM, Carlos Chavez wrote: I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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] Has anyone sucessful Asterisk to an Avaya IP phone
Bob, We are on a similar assignment right now. Please contact me off-list if you would like to discuss how we might be helpful. Thanks, Bryan M. Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Bob Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 12:25:15 PM (GMT-0500) America/New_York Subject: [asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. [EMAIL PROTECTED] - -- We've Got Your Name at Mail.com Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] Headset for Polycom
With Polycom phones, you should steer clear of headsets with in-line amplifiers. We have found these to introduce electrical hum into the audio streams. Just an FYI. Thanks, Bryan Johns Partner Shelton | Johns 1805 Old Alabama Road Suite 200 Roswell, GA 30076 USA Office: 678.248.2637 FindMe: 678.229.1809 Email: [EMAIL PROTECTED] On May 4, 2007, at 11:15 AM, Mike wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike ___ --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 ___ --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 1.4.3 segfaults on receiving calls.
We saw this behavior early in the 1.4 releases and shelved 1.4 upgrades for the time being. The behavior that we saw was similar to what you describe. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 25, 2007 12:57:54 PM (GMT-0500) America/New_York Subject: [asterisk-users] Asterisk 1.4.3 segfaults on receiving calls. On upgrading 2 machines (1 with a very simple configuration) from asterisk 1.4.2 to 1.4.3, I have found that on receiving a call (on either an IAX2 or SIP channel) the server process segfaults. Is anyone else having this trouble? ___ --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 ___ --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] Random Asterisk deaths
What version are you running? Anything creative like VMs or other unique configurations in use? Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Wayne Jensen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 24, 2007 7:24:26 PM (GMT-0500) America/New_York Subject: [asterisk-users] Random Asterisk deaths Every once in a while for no apparent reason, Asterisk has been dying on me, dropping all calls in progress. There's nothing in the log file or on the Asterisk console that indicates the reason. Some days it doesn't happen at all. Other days it happens two or three times. The problem began on Friday, but the last time anything was changed on that box was at least a week before that. Any suggestions on what to do/where to look to find out what's going on and fix the problem? ___ --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 ___ --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] Polycom SIP Phones On LAN can't register without WAN (Internet) Access
Might want to confirm what server address you have declared in your sip.cfg file (assuming you are using network provisioning for the phones). Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, April 23, 2007 12:48:25 PM (GMT-0500) America/New_York Subject: Re: [asterisk-users] Polycom SIP Phones On LAN can't register without WAN (Internet) Access Noah Miller wrote: Hi Shawn - We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server. I've checked everything I can think of but can't figure out why its happening. I believe since the Asterisk Box is on the LAN and the phones are on the same LAN then it shouldn't need internet to function. The closest I have narrowed this down is to the DNS area. I found that if I block access to our ISP's DNS that the phones aren't able to register with asterisk. This baffles me because the phone has the LAN address for the Asterisk server so I don't know why it's doing DNS lookups. Hmm. Well, you've got me. I don't know why it would be doing that, it certainly shouldn't be. You might try a newer version of the SIP firmware or the 3.2.2 bootrom. If it still happens with the latest bootrom/firmware, you could do a packet trace on the phone. Is it doing DNS queries? If so, I'd call your Polycom reseller and have them take this up with Polycom (support requests are supposed to go through the reseller). Actually, in any case, I'd take it up with your Polycom reseller. Asterisk tends to get very upset when DNS is down. Make sure you have NO hostnames in any of your Asterisk config files. Also make sure that all interfaces on the Asterisk box are correctly listed in /etc/hosts ___ --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 ___ --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] No of Calls
Install zaptel and only enable the ztdummy module. As long as you are not running in a VM, this will supply you the timing that you are looking for. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Arun Kumar [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Thomas Kenyon [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 4:54:47 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] No of Calls how do I check that whether trunking is working or not ? No I don't any timing soure (like zaptel card) b'coz these are test server. what else I can use for timing. thanks On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Arun Kumar wrote: I've tried this but stil some problem Like if I use this link that you gave me it shows for 10 call 136.08KBps in one direction, but, when I place call using my phone for 10 calls it comes 210KBps in one direction. Ar eyou sure trunking is working? Do both asterisk servers have a timing source? ___ --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 ___ --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] Recommendations for a voip provider who supports LNP?
Have a look at these guys: http://www.vitelty.com I have had good success with their service (particularly with porting). Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Salvatore Giudice [EMAIL PROTECTED] To: Baji Panchumarti [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 9:35:12 PM (GMT-0500) America/New_York Subject: RE: [asterisk-users] Recommendations for a voip provider who supports LNP? I need a straight origination/termination provider on a per minute charge plan. I would like to avoid a monthly subscription-based provider. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, April 17, 2007 6:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Recommendations for a voip provider who supports LNP? On 4/17/07, Salvatore Giudice wrote: (sorry about the repost. I accidently had an unrelated subject in the original) Can anyone recommend a VoIP provider who supports LNP? I need to move to a new provider for inbound calling and I want to keep my current numbers. My current provider is a gaggle of retards. Any recommendation? I need a service that is reliable. TIA, SG have you considered teliax.com ? check your numbers for LNP at the bottom left. I have been playing with voip for only about a month, but no complaints with teliax svc so far. -baji. -- ___ --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 ___ --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 ___ --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: Re[2]: [asterisk-users] openvz resources
What you are describing is only available in a guest domain if your CPU(s) support hardware virtualization. If they do, however, this configuration is pretty straight forward. Xen as a virtualizing solution ships in a well-documented format in the Fedora 6 distribution. If you would prefer to run it in etch, you should dig into the docs available from http://www.xensource.com. I hope that this is helpful. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Gunnar Schaller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, April 14, 2007 5:57:28 AM (GMT-0500) America/New_York Subject: Re[2]: [asterisk-users] openvz resources Hello, Can you tell more about Xen? I would like to install Debian Etch with Xen and use A Digium 4-port E1 in a guest domain. Is it possible? I read of much problems with cards in a guest domain. I have Xen running with DNS-server/ Web-server guests, also a VoIP only Asterisk, but a telephony card is missing in a guest. Gunnar Schaller Saturday, April 14, 2007, 1:01:07 AM, you wrote: No relevant experience with OpenVZ, but plenty with Xen if you would find that interesting. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com ___ --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 ___ --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] SpanDSP (RxFax)
Might want to look into you libtiff version and / or the presence of tiff2pdf. Just a guess. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Sahil Gupta [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 13, 2007 5:27:04 PM (GMT-0500) America/New_York Subject: [asterisk-users] SpanDSP (RxFax) Hi, We had an install working quite well of SpanDSP on our machine until recently where it has began spitting out an error stating unable to translate from unknown to unknown. Any ideas ? Regards, Sahil Gupta VoiceValley ___ --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 ___ --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] openvz resources
No relevant experience with OpenVZ, but plenty with Xen if you would find that interesting. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com - Original Message - From: Voip Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, April 13, 2007 6:04:16 PM (GMT-0500) America/New_York Subject: [asterisk-users] openvz resources Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles ___ --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] FAX thru TDM400p
Is your carrier delivering service via a TDM circuit? It has been our experience that you will get far more reliable fax performance via the method you describe (analog device terminated to a port on a FXS line card) than attempting to use an ATA on the LAN. However, if your carrier is a SIP or IAX trunking provider, your reliability concerns are on the other side of your SIP switch. Bryan Johns Partner Shelton | Johns 1805 Old Alabama Road Suite 200 Roswell, GA 30076 USA Office: 678.248.2637 FindMe: 678.229.1809 Email: [EMAIL PROTECTED] On Apr 6, 2007, at 8:39 AM, Joe Acquisto wrote: There seem to have been many discussions about this, so sorry if this is boring. Can one connect a standard fax machine (or fax modem) to an analog port on a TDM400p (as if it were an analog phone, say) and expect it to work reliably? For sending, that is. Detecting and routing the call is another subject (for me). Seems it should, but does not. At least not for me. joe a. ___ --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 ___ --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] Linksys/Sipura SPA-942 phones in larger deployments
Chris, Having deployed every major brand of sip handset in numbers greater than 100, I can say that I recommend the Polycom product hands-down for these types of roll-outs. Provisioning and management are superior and the product if of generally better quality than the SPA line. If you want more details, please feel free to contact me off-list. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Mar 22, 2007, at 3:50 PM, Chris Bagnall wrote: Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/ configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with a similar feature set and price range? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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 ___ --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] Linksys/Sipura SPA-942 phones in larger deployments
Dave, I highly recommend that you try network provisioning the Polycom phones you have. The configuration access and tweaks available in the config files is nearly infinite and can be used to address most, if not all of the issues that you mention here. Here's a decent run-down on network provisioning the Polycom: http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501 While it may not be practical to go this route for a single desk handset, it can be a life-saver in a larger network rollout. I hope that this helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Mar 22, 2007, at 6:51 PM, dave cantera wrote: bryan, can it be that polycom is the best? my headset doesn't work with the 600 volume resets to default on every hangup, speakerphone resets intermittenly (haven't figured out why), my 301 has a speaker phone but no mic (very useful!), every config changes reboots the phone taking 60+ seconds to restart, if you change 'Registration 4' *and* 'Registration 5' only the submit button you clicked on updates, the other registration remains the same... granted, I haven't tried the server tftpboot option, but even so, making minor changes is quite a chore... perhaps once you hack though it and things settle out, it gets better?... please help me in my miss-understanding of polycom, I am still looking for a good sip phone... maybe you could convince me to give them another chance? thanks for your help. daveC Bryan M. Johns wrote: Chris, Having deployed every major brand of sip handset in numbers greater than 100, I can say that I recommend the Polycom product hands-down for these types of roll-outs. Provisioning and management are superior and the product if of generally better quality than the SPA line. If you want more details, please feel free to contact me off-list. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Mar 22, 2007, at 3:50 PM, Chris Bagnall wrote: Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with a similar feature set and price range? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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 ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.15/728 - Release Date: 03/20/2007 08:07 AM -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --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 ___ --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] Polycom Questions
DST rules can be found by searching the sip.cfgg file for SNTP. You will find a cluster of time parameters, including the month and day upon which to change DST. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Mar 5, 2007, at 9:20 PM, »Steven Ringwald« wrote: Any Polycom gurus out there? If so, I have a few config file questions. First off, does anyone have the daylight savings time rules written for this Sunday's big change? Secondly, if there any way in the config file to tell the phone not to display the number of missed calls? I don't mind it keeping the missed calls list, I just don't want that running count. Lastly, I am trying to get the dialplan to work, but have had no luck so far. I have tried defining it in the sip.cfg and/or the phone1.cfg, but have had no luck getting the phone to latch onto the numbers, and immediately dial. I am running with the 2.0.1 firmware, if that matters. from sip.cfg: dialplan dialplan.impossibleMatchHandling=0 dialplan.removeEndOfDial=1 digitmap dialplan.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx[2-9] xx dialplan.digitmap.timeOut=3/ routing server dialplan.routing.server.1.address=10.0.17.8 dialplan.routing.server.1.port=5060/ emergency dialplan.routing.emergency.1.value=911 dialplan.routing.emergency.1.server.1=1/ /routing /dialplan from phone1.cfg: dialplan dialplan.1.impossibleMatchHandling=0 dialplan. 1.removeEndOfDial=1 dialplan.2.impossibleMatchHandling=0 dialplan.2.removeEndOfDial=1 dialplan. 3.impossibleMatchHandling=0 dialplan.3.removeEndOfDial=1 dialplan.4.impossibleMatchHandling=0 dialplan. 4.removeEndOfDial=1 dialplan.5.impossibleMatchHandling=0 dialplan.5.removeEndOfDial=1 dialplan. 6.impossibleMatchHandling=0 dialplan.6.removeEndOfDial=1 digitmap dialplan.1.digitmap=9[2-9]xx[2-9]xx|91[2-9]xx [2-9]xx dialplan.1.digitmap.timeOut=3 dialplan.2.digitmap= dialplan.2.digitmap.timeOut= dialplan.3.digitmap= dialplan. 3.digitmap.timeOut= dialplan.4.digitmap= dialplan. 4.digitmap.timeOut= dialplan.5.digitmap= dialplan. 5.digitmap.timeOut= dialplan.6.digitmap= dialplan. 6.digitmap.timeOut=/ routing server dialplan.1.routing.server.1.address=10.0.17.8 dialplan.1.routing.server.1.port=5060 dialplan.2.routing.server. 1.address= dialplan.2.routing.server.1.port= dialplan. 3.routing.server.1.address= dialplan.3.routing.server.1.port= dialplan.4.routing.server.1.address= dialplan.4.routing.server. 1.port= dialplan.5.routing.server.1.address= dialplan. 5.routing.server.1.port= dialplan.6.routing.server.1.address= dialplan.6.routing.server.1.port=/ emergency dialplan.1.routing.emergency.1.value= dialplan. 1.routing.emergency.1.server.1= dialplan.2.routing.emergency. 1.value= dialplan.2.routing.emergency.1.server.1= dialplan. 3.routing.emergency.1.value= dialplan.3.routing.emergency. 1.server.1= dialplan.4.routing.emergency.1.value= dialplan. 4.routing.emergency.1.server.1= dialplan.5.routing.emergency. 1.value= dialplan.5.routing.emergency.1.server.1= dialplan. 6.routing.emergency.1.value= dialplan.6.routing.emergency. 1.server.1=/ /routing /dialplan Thanks in advance! Steve ___ --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 ___ --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] HELP!! Dropping calls on Bridge
What asterisk version? Bryan M. Johns Partner Shelton Johns Technology Group Office: 678.248.2637 Direct: 678.229.1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Jason Wolfe [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 2/21/2007 11:01 AM Subject: [asterisk-users] HELP!! Dropping calls on Bridge All calls through the system are being dropped when they are bridged (Asterisk, Linux, pure VoIP system). The calling party here's half of the word 'hello' for instance and the call is dropped. I've noticed that hangup() was being called for some time now when the call was bridged, but the call was still continuing. Any thoughts on where to start debugging? Jason ___ --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 ___ --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] Best phone for easy provisioning
I can only speak for Aastra phones. Central provisioning is very easy. All you need is one simple text file on a TFTP, FTP, or HTTP server which all the phones point to. To customize individual phones you add a second text file for each phone you want customized. The custom text file is given the name of the phones MAC address. When the phone reboots it first reads the general text file and then reads it's custom file which will overwrite any duplicate setting in the general text file. To remotely reconfigure and reboot phones you can configure them to check for updates to these files or for updated firmware at a certain time of day. You can also remotely reboot individual phones based on extension from the Asterisk CLI. Of course, you can also access an individual phones Webpage configuration based on it's IP address. From: Rod Bacon [mailto:[EMAIL PROTECTED] Sent: Thursday, February 08, 2007 12:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best phone for easy provisioning Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I’ve used CISCO 79XX phones, and they’re great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? Rod Bacon Technical Manager JASCO Consulting Pty. Ltd. http://www.jasco.net.au Ph. 03 9432 6376 Fax: 03 9432 6378 Polycom's central provisioning is very straight forward and very powerful. There is support for all major connectivity methods (tftp, ftp, ftps, http, https, etc) and the configuration capability is more broad than any other phone we have worked with. I hope this information is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com ___ --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] Help - Poor Voice Quality
Run mtr on your server against the registration server at Teliax and look for bad hops on your route to and fro. If you don't find anything there, you may want to fire up ethereal and capture packets on a few calls and look through them for error data that may be contributing to bad voice quality. I hope this is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Feb 6, 2007, at 8:09 PM, Jim Duda wrote: I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with QOS configured for VOIP P1 o Asterisk 1.4.0 o Sipura SPA-2000 o Grandstream GXP-2000 o IAX connection to teliax Outbound voice quality is many times horrible, to the point where ppl say they cannot hear me. The voice often drops out. Inbound quality seems to cut in and out too. I downloaded the myVoipSpeed VOIP analyzer. It indicates that I have plenty of download and upload bandwidth. I also have good jitter. The tool doeesn't find any packet loss whatsoever. My RCN cable company cannot find anything wrong with my cable modem. No packet loss. I'm supposed to be paying for 10M bit downloads, but only getting 3M bit. I've been on the shorewall firewall and confirmed that I have the firewall configured properly for VOIP QOS. I'm using the basic asterisk iax.conf setup with only those changes required to interface with the teliax service. I have the same issues with both the Sipura Adapter and the Grandstream phones, however, I do believe the Grandstream appears worse at times. I've attempted to analyze the IAX traffic using the Wireshark ethernet protocol analyzer. Everything looks okay best I can tell. What else can I do to analyze why the voice quality is so bad? What can I do in Asterisk to help track down where the problem is? I want to make this VOIP work. Thanks for any help. Jim ___ --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 ___ --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] Buddy list order
Assuming you are using a central provisioning server, check your
{MAC}-directory.xml file. It contains the ordering that you are
looking for.
I hope this helps.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Feb 6, 2007, at 9:35 PM, Bill Gibbs wrote:
I could have sworn I saw a post about this recently but I can’t
find it so apologies if this is a dupe, but is there anyway to
control the order in the Polycom Buddies list?
Bill
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Re: [asterisk-users] Polycom Provistioning Issue
Jason, Email me off-list and I will ship you a pack of usable configs. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 26, 2007, at 3:48 PM, Jason Walker wrote: Fixed that issue but it does not change the error 0126204105|cfg |3|00|Image sip.ld has not changed 0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 0126204105|cfg |3|00|Downloaded application image is identical to current version 0126204105|cfg |3|00|Phone successfully provisioned 0126204136|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0126204136|app1 |6|00|Uploading boot log, time is FRI JAN 26 20:41:36 2007 William M. Conlon wrote: Looks like the network time server isn't provisioned. -- Bill 1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0). ___ --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 ___ --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 ___ --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 1.4 Polycom buddy status
I ran into this problem with an early batch of IP650s. Polycom's firmware version 2.0.3b made this issue go away. Thanks, Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 23, 2007, at 10:09 AM, Chris Bullock wrote: I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will stick in the busy status. I have noticed that I can call that extension the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris ___ --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 ___ --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] Polycom registration fails
Are you using tftp or ftp provisioning? If so, check your server declaration in sip.cfg in your polycom configs directory. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 14, 2007, at 2:54 AM, Al wrote: Hello list, I was wondering if any of you guys have had any luck with polycom in remote offices, I'm facing a weird issue, polycom phones work fine in the main office, in remote office it says, Registration from 'sip:[EMAIL PROTECTED]' failed for '70.59.21.112' - Wrong password the odd thing is Linksys phone works without any issue!! polycom wont register but its able to place calls!!! in wiki it says: If the phones fail to register with Asterisk but can still make outbound calls, you likely need to adjust the digest realm parameter from the default of PolycomSPIP.(http://www.voip- info.org/wiki-Polycom%20Phones) anyone knows what does it exactly mean? ___ --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 ___ --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] Directory too difficult?
I wish had some pearl of wisdom here, but I don't. I will simply share my sympathy. Sounds like an ESU situation to me. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote: I have a group of users whos complaint about Asterisk is that the directory application is too hard too use. (yeah, yeah, I know. For the record, they're Calgarians) Now I'm in a pickle: I don't want to have to create a custom directory for these guys. Anyone have any tips for making the directory easier, maybe re-record the prompts so they are more verbose? We go by first name. ___ --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 ___ --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] Directory too difficult?
Exactly. ESU = Equipment Superior to Users ;-) Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote: More like a ID-10-T error….. From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Bryan M. Johns Sent: Wednesday, January 10, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Directory too difficult? I wish had some pearl of wisdom here, but I don't. I will simply share my sympathy. Sounds like an ESU situation to me. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote: I have a group of users whos complaint about Asterisk is that the directory application is too hard too use. (yeah, yeah, I know. For the record, they're Calgarians) Now I'm in a pickle: I don't want to have to create a custom directory for these guys. Anyone have any tips for making the directory easier, maybe re-record the prompts so they are more verbose? We go by first name. ___ --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 ___ --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 ___ --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] Voicemail personalised greetings using DB/IMAP backend?
We are in a project right now where we have build a single asterisk switch acting as a master SIP router and delivering service to and from about 30 xen-based VMs. It is a multi-tenant build. I am not certain if this is your particular scenario or if I am off-base. A word of caution, though. Do not run SIP routing functions on Dom0 in a Xen environment and do not use Asterisk 1.4 for these functions yet. In testing, we encountered routine segmentation faults on both our Dom0 and our 30 DomUs. We fixed this issue by separating the core SIP routing functions to a stand-alone server and by downgrading all DomUs to Asterisk 1.2.14. Our entire architecture is Fedora 6, by the way. DomU is 32bit and all DomUs are run on a single, large 64bit server platform. I hope this is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 5, 2007, at 7:00 PM, Ray Jackson wrote: Hi Bryan, I was trying to avoid creating an architecture dedicated to VM, but have Asterisk handle VM in a horizontally scalable way. I understand there are some issues with MWI etc. if you separate out the VM from Asterisk? Could you point me at any good examples of a VM architecture I could use as a reference? Cheers, Ray Bryan M. Johns wrote: Ray, Have you considered using a VM architecture? Bryan M. Johns Partner *Shelton | Johns Technology Group* office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 *http://www.sheltonjohns.com* http://www.sheltonjohns.com/ On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote: Hi all, I am attempting to build a horizontally scalable Asterisk deployment and am getting very close to achieving that goal. With Asterisk 1.4 I now have an IMAP backend for Voicemail messages which is great as users can check the same messages either through the voice portal or using Webmail. However, I'm not sure the best way of dealing with personalised greetings such as a user's unavailable/busy message etc. Despite the IMAP backend these greetings appear to be stored on the local file system under /var/spool/asterisk/voicemail/default, which means if I build a farm of Asterisk servers - each will have it's own spool directory. My aim is to have *nothing* stored locally at all... If there a way of storing these greetings in a database table or using IMAP? I saw the ODBC voicemail storage module, but I would prefer to stick with a REALTIME/IMAP backend? If I mount the / var/spool/asterisk/voicemail directory remotely using a shared NFS mount on a NAS device will this work okay or lead to problems/ race conditions etc.? Any advice would be welcome! Regards, Ray ___ --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 - --- ___ --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 ___ --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 ___ --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] Voicemail personalised greetings using DB/IMAP backend?
Ray, Have you considered using a VM architecture? Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote: Hi all, I am attempting to build a horizontally scalable Asterisk deployment and am getting very close to achieving that goal. With Asterisk 1.4 I now have an IMAP backend for Voicemail messages which is great as users can check the same messages either through the voice portal or using Webmail. However, I'm not sure the best way of dealing with personalised greetings such as a user's unavailable/busy message etc. Despite the IMAP backend these greetings appear to be stored on the local file system under /var/ spool/asterisk/voicemail/default, which means if I build a farm of Asterisk servers - each will have it's own spool directory. My aim is to have *nothing* stored locally at all... If there a way of storing these greetings in a database table or using IMAP? I saw the ODBC voicemail storage module, but I would prefer to stick with a REALTIME/IMAP backend? If I mount the /var/ spool/asterisk/voicemail directory remotely using a shared NFS mount on a NAS device will this work okay or lead to problems/race conditions etc.? Any advice would be welcome! Regards, Ray ___ --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 ___ --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] caller id ring tones for Asterisk Phone
Most SIP phones handle this functionality by recognizing numbers from speed dial or address book entries in the phone itself. I believe that the PolyCom SIP phones do this (IP430, IP501, IP601, IP650). I hope that this is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Jan 3, 2007, at 11:15 PM, Jeronimo Romero wrote: I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!! ___ --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 ___ --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] Realtime multiple registration for a HardPhone Snom 360 (solved)
I guess I misunderstood your issue, Fred. Have a great New Years. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com On Dec 30, 2006, at 8:59 AM, Asterisk [Submusic] wrote: Hi all, My problem seems to be solved, When we have multiple SIP accounts on the same phone with RealTIme configuration, Asterisk can't authenticate correctly the second account, I think it's because of the same IP and port number. My solution is to use insecure=invite on the second SIP account in the database. Thanks for your answer Bryan, but I don’t like FreePBX, I prefer VI :-) Fred -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Bryan M. Johns Envoyé : vendredi, 29. décembre 2006 15:58 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [asterisk-users] Realtime multiple registration for a HardPhone Snom 360 The device config for the Snom 360 needs to be set to adhoc mode. If you are not comfortable with hand-configuration of the extensions file, take a look at freepbx as a tool to assist you. Thanks, Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Frédéric Marti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 12/29/2006 9:25 AM Subject: [asterisk-users] Realtime multiple registration for a Hard Phone Snom 360 Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --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 ___ --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 ___ --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] Realtime multiple registration for a Hard Phone Snom 360
The device config for the Snom 360 needs to be set to adhoc mode. If you are not comfortable with hand-configuration of the extensions file, take a look at freepbx as a tool to assist you. Thanks, Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Frédéric Marti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 12/29/2006 9:25 AM Subject: [asterisk-users] Realtime multiple registration for a Hard Phone Snom 360 Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --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] Re: [OT] Wifi SIP phones - LinkSys WIP330
I recommend the hitachi wifi phones for use with asterisk. Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Steven [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 12/28/2006 4:30 PM Subject: [asterisk-users] Re: [OT] Wifi SIP phones - LinkSys WIP330 I bought a WIP300 to test and it was aweful. It would either not register a keypress or register it twice. It would also freeze up few minutes at a time. It looks like the WIP330 has a new keypad, so maybe that problem is gone. The WIP300 worked with asterisk, but I can not recall the quality at this point. -- -- Steven http://www.glimasoutheast.org Wayne [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne . ___ --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 ___ --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 ___ --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] Re: Help with silence or gating of speech?
On Dec 22, 2006, at 3:56 AM, Tony Mountifield wrote: In article [EMAIL PROTECTED], Robert Jenkins [EMAIL PROTECTED] wrote: Hi, I'm using Asterisk (1.2.13) on Centos 4.4 x86_64 with a TDM2400E for analog trunks ( extensions) plus some Polycom 501 601 phones. I have a problem in that the audio via the Polycoms is gated or muted during quiet parts of the other person's speech. I'm not familiar with the Polycoms, but I would guess that they have some kind of web-based setup screens. Look for silence suppression or VAD (voice activity detection) and set it to Off or Disabled. Otherwise you could try visiting http://bugs.digium.com/view.php? id=5374 and apply the patch 2005-10-04-3-asynchronous.patch It's the patch to channel.c that is the important part. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 The PolyCom handsets have lots of tune-able audio parameters that are manipulated using the phone's configuration file. Assuming that you are using TFTP or FTP config distribution, these files are the [phonemacaddress].cfg file and the master sip.cfg file. I recommend looking through the settings available in these files and maybe doing some research on the config of particular PolyCom models at http://www.voip-info.org. Backup your original configs before making edits, though ;-). I hope this is helpful. Bryan M. Johns Partner Shelton | Johns Technology Group office: 678:248:2637 x:1500 direct: 678:229:1809 mobile: 404.259.9216 iaxtel: 700:248:2637 x:1500 http://www.sheltonjohns.com ___ --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] Polycom 500 Sound Problem
What DTMF mode are you using? ___ 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: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ 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
Re: SV: [Asterisk-Users] Presence and IM?
Bjorn, Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal It has a flash-based panel that will give you what you are looking for. Bryan M. Johns One Ring Networks 300 West Wieuca Road, NE Building One Suite 205 Atlanta, GA 30342 404.303.9900 x: 104 http://www.oneringnetworks.com On Fri, 2005-06-17 at 20:33 +0200, Bjorn wrote: Maybe, but that would not have been a reliable way of handling it, as not all users would necessarily use voicemail. Besides, I would think that this feature is supported by several SIP devices (it has to do with messaging), so it would be better If Asterisk supported this feature by default, no hacking needed. Regards, Bjorn Fra:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] P vegne av Dean Collins Sendt: 17. juni 2005 18:11 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Presence and IM? Hi Bjorn, Maybe it could be done as some form of check against call forward to voicemail etc. Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bjorn Sent: Friday, 17 June 2005 11:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Presence and IM? We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se whos online and available and whos not. Surely, theres the manager interface, but unless youd want to install extra software on each client computer, this is not a good option. Then theres the presence / IM function in SIP. Since were only using SIP clients, this could easily solve some of our problems. However, I cannot get this to work with Asterisk using Eyebeam. Is this because the function is simply not supported within Asterisk? If lack of support is the case, anyone knows if this feature is to be implemented in the near future? Regards, Bjorn ___ 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
[Asterisk-Users] Routing SIP to Cisco routers running IOS 12.3+
I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as available or registered. There is little to lead me to believe that IOS will support a port-over-port SIP registration with Asterisk, so I have configured sip_additional.conf with the following format for each extension on the 1751: [XX] username=XX type=friend port=5060 nat=yes host=xxx.xxx.xxx.xxx dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=XXX XXX allow=alaw I am relatively confident that my problem does not exist at the 1751 due to the ability to flawlessly process outgoing calls. However, after more than a day in this one, I guess anything is possible. Does anybody out there have any experiencing sending SIP down-wire from Asterisk to the Cisco IOS? We might be willing to pay for the right kind of help here. Thanks. ___ 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] Dial Commands D Option Question
On Thu, 2005-06-16 at 23:24 -0400, Nate Kapi wrote: When using the dial command and the D option to send DTMF digits when the channel is answered, is there a way to allow for some dead air, and then send more DTMF digits? I would like to automate a call, and it requires entry of a few short dtmf digits all a couple seconds apart from each other. Thanks! ___ 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 >From a thread in February of this year: At least in app_dtmf_stream(), it's just hard coded in there as 100 or last argument to app_dtmf_stream(). Nick Hope this is a start. ___ 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
