[asterisk-users] dial tree crawler?
For lack of a better term, I have been tasked with creating a dial tree crawler. The reason is that we have a soon to fail Octel system. The major issue is that there is no way to port the dial tree recordings from the Octel. So what I envision is creating a script that can somehow dial down the trees spawning a recorded call for each tree. I realize that it will take time to cut these down and tidy them up but it will be a lot less work than recreating thousands of trees for sure. So I would need to figure out some way to realize what the options are and dial pass the DTMF to get into the subtree. Anyone out there ever had to do something similar or have any suggestions on how to accomplish this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RF to IP bridge
I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Thanks! Curt ___ --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] RF to IP bridge
Yeah...I'm a HAM myself so I have used auto patch. This is on a larger level. I'm looking at something like http://www.twistpair.com/ which we are about to implement in our enterprise level. However I'm looking for a cheaper alternative and one that works with Asterisk for us HAMs to have some fun with :) RoIP is the buzzword for it I suppose. Looks like there is some good potential there somewhere for the open source world. Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Per Jessen Sent: Thursday, May 31, 2007 3:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] RF to IP bridge Curt Shaffer wrote: I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Radio-amateurs have done phone-patching for decades (where allowed) - there must be someone who can point you in the direction of an easy solution. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] RF to IP bridge
Half duplex is not an issue. Basically the idea is radio over IP. I don't want to change the fact that we are using radios. For example, on an enterprise level I'm going to be working with a crew to set this up for our Avaya system. It is basically for emergency communications. Say the fire chief is out of town and something major happens. We would like for him to be able to call in and hear and interact with the squad on site via the radio network from the PSTN or even a cell phone. With http://www.twistpair.com/ this is completely possible but that only integrates with Avaya or Cisco Call Manager at this time. Not a problem as we run Avaya on an Enterprise level but I'm looking for free or cheap alternatives. Another example and more towards what I am looking at. As a RACES (Radio Amateur Civil Emergency Service) member I would like to have a crash cart that would allow instant ability for communications on a range of mediums. GSM cards, EVDO, WIFI, and radio communications all from a small box that can be very mobile and run on something like a gel cell batteries. The ability to bridge between the two would be very useful in cases of disperse conditions where every RACES member could be offering communications to victims outside of net repeaters or have another medium to get back into the tactical net rather than having to utilize repeaters out of the range of the net control. We have internet controlled repeaters and utilize VoIP on a lot of them but we are looking for something that can be small, very mobile and offer other services other than just radio communications. And just FYI the ~$200/channel is for the above named software that does just what I'm explaining. Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Coccimiglio Sent: Thursday, May 31, 2007 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RF to IP bridge Per Jessen wrote: Radio-amateurs have done phone-patching for decades (where allowed) - there must be someone who can point you in the direction of an easy solution. /Per Jessen, Zürich The BIG problem here is that most Radio Amateur software and hardware operate in a half-duplex manner. I don't think that would be what you want. If half-duplex is ok then most radio makers (Icom, Motorola, etc.) have complete turn-key solutions. If you want it cheap then your will have to build it yourself. I don't see $200/channel happening in either case for VHF/UHF. Please share more info and maybe I can help. Mark C ( N3WHX ) [EMAIL PROTECTED] sip:[EMAIL PROTECTED] (VoIP) ___ --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] RF to IP bridge
This looks kind of along the lines of what I'm looking for! I will explore it's abilities. Thanks! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shane Young Sent: Thursday, May 31, 2007 6:03 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] RF to IP bridge Quoting Curt Shaffer [EMAIL PROTECTED]: I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Asterisk does this quite well: wawnmnxast1*CLI core show application Rpt wawnmnxast1*CLI -= Info about application 'Rpt' =- [Synopsis] Radio Repeater/Remote Base Control System [Description] Rpt(nodename[|options]): Radio Remote Link or Remote Base Link Endpoint Process. Not specifying an option puts it in normal endpoint mode (where source IP and nodename are verified). Options are as follows: X - Normal endpoint mode WITHOUT security check. Only specify this if you have checked security already (like with an IAX2 user/password or something). Rannounce-string[|timeout[|timeout-destination]] - Amateur Radio Reverse Autopatch. Caller is put on hold, and announcement (as specified by the 'announce-string') is played on radio system. Users of radio system can access autopatch, dial specified code, and pick up call. Announce-string is list of names of recordings, or PARKED to substitute code for un-parking, or NODE to substitute node number. P - Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. For the user to have DTMF control, the 'phone_functions' parameter must be specified for the node in 'rpt.conf'. An additional function (cop,6) must be listed so that PTT control is available. D - Dumb Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. In this mode, the PTT is activated for the entire length of the call. For the user to have DTMF control (not generally recomended in this mode), the 'dphone_functions' parameter must be specified for the node in 'rpt.conf'. Otherwise no DTMF control will be available to the phone user. ___ --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
[asterisk-users] Ser vs. DUNDi
With all of the recent talk on the list about DUNDi, I have a question. From the outset it appears that SER is often used for high availability solutions and as a tool for almost clustering Asterisk boxes behind it. It appears to me that DUNDi is providing a lot of this as well. Now I know DUNDi is not an application by itself to proxy SIP requests but can I hear any information out there that supports that DUNDi is in fact a valid alternative to something like SER or not? A nice feature analysis between the two in a clustering/highly available solution would be nice to see. Not a feature list but rather a discussion from people that have tested/used both for people who are deciding which way to go to achieve the goal. Thanks! Curt ___ --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] intermittent choppy sound over wifi link
I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider - IAX2 over the Internet - 20Mb fiber connection - router - Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running 3.1.19(SE) firmware), also tested with X-lite softest - PIX 506 (although I have tried multiple routers and direct connection to the radio try to fix the problem) - 1 mile 802.11b link to AP - 15 mile 802.11b link Backhaul - router - Asterisk My Asterisk version is Asterisk 1.2.12.1, Zaptel 1.2.9.1. Ping times are ~10ms, jitter is under 10 with an average of 5. QoS is enabled in the router for SIP, RTP and IAX2 traffic going to and from the Asterisk box. When I experience the choppiness the ATA reports packet loss on the web interface (Call 1 Packets Lost: ). I can run something such as ping plotter from the same leg of the network that the Asterisk box is on while this is happening and there is not even a small glitch of lost packets on the network but the ATA displays otherwise. The only thing I have come up with thus far is possible retransmissions on the wireless connection (and due to the type of gear, I'm not able to see this data). We are way out in the country with no other real providers even close so I'm doubting interference although I suppose it is a possibility keeping an open mind. My question is can anyone point me to any possible reasons this would be happening? Also can anyone tell me other reasons other than real lost packets that the ATA would show this? My only guess on that was packets that never got an ACK due to server congestion or some other reason other than actual loss. Any insight appreciated! Thanks Curt ___ --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] Cisco + Asterisk list anyone?
I have been working with a couple companies who are interested in integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk for AA, VM, failover trunks etc. I have found some materials and guidance out there but I think a list and/or wiki for general asterisk integration with other vendors would be great and feel that it is enough off topic that it deserves its own space. Just throwing it out there for feedback. I'm willing to host both. Let me know what ya'll think! Curt ___ --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] OT: Sipura DST Rules
Thanks a million! Just verified after putting it in my encrypted configs and it works like a charm! :) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Monday, March 12, 2007 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Sipura DST Rules Since we've had discussion about DST on polycom I thought I'd pass along the rule I used to configure DST on my sipura units as well (This way the date and time passed in caller ID will be correct). Under the admin view go to the regional tab. At the bottom under miscellaneous enter this in Daylight Saving Time Rule: start=3/8/7/2:0:0;end=11/1/7/2:0:0;save=1 This is based off information I found here: http://www.sipura.com/Documents/faq/Section_5.html -Dave ___ --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] To use asterisk or proprietary hardware, that is the questio
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Totaro Sent: Monday, February 26, 2007 11:11 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that is the questio From: shadowym [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [asterisk-users] To use asterisk or proprietary hardware,that is the question Date: Mon, 26 Feb 2007 20:42:21 -0800 Thanks Tom and everyone else, Based largely on your comments I decided to just stick with what works. I have a site using entry level ATX server hardware that has been solid as a rock. I'll just go with that instead of more specialized fanless hardware, specialized power supply and 2.5 hard drives etc. Maybe get a second motherboard as a spare of they go for the ongoing remote support option. I'll do some simple things like a put in a standby hard drive with the production image on it in case the primary drive fails. The case has hot swap SATA bays so if the primary drive fails or get's corrupted anyone can just swap drives and they will be back up just like that. I'll make remote offsite backups as well. Thanks for all the help. -Original Message- From: Tom [mailto:[EMAIL PROTECTED] Sent: Saturday, February 24, 2007 5:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] To use asterisk or proprietary hardware,that is the question At 11:53 AM 2/24/2007, you wrote: Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a set it and forget it type of thing like a lot of small business proprietary systems. There is no such thing as set and forget. Businesses change. They either grow or shrink, they don't stand still. They will add and remove phones. So they will call you at that time. Or are you expecting them to shop for their own phones on Ebay? At the same time they would like to be able to do minor dial plan changes themselves so I have determine that a GUI like FreePBX or similar alternative (free or commercial) is appropriate. We take a different approach. We don't want a GUI. We don't want the limits. We work with the business to design their dial plan. Then we write it. We do not give them a GUI because we don't want them making changes and then asking for support. We sell them a minor service agreement and remote in for any changes. We also handle professional voice recording and basic training on phone use. And we handle backups and service if needed. Once they understand that we can do that without a service call, they are quite receptive to the idea. Conventional PBXs come with service agreements so customers are used to that but surprised at the low cost from you. I have some concerns about using Asterisk for this. As much as I am in support of the whole Asterisk revolution, I just do not feel confident enough in Asterisk on a Hard Drive as a set it and forget it setup running month after month, year after year. I am hoping someone can convince me otherwise. Hard drives are reliable. But I have similar feelings so we are working on a flash solution. Were running it beta in our office right now. It only uses the hard drive for daily voicemail, boots from flash and runs from RAM. I'm concerned about hard drive corruptions/failures, memory leaks, software bugs etc. Conventional systems have bugs too. I have the budget to buy good quality hardware so if I was to go with Asterisk I would go industrial grade fanless computer, power conditioned UPS etc. You don't really need fanless. Make it cheap enough that it can easily be replaced. Like a $500 PC. I am not concerned about the reliability of most of the hardware. It's the hard drive and the software that runs on it that worries me. I will obviously use a mature stable Asterisk release and the most stable Linux version which I won't bother naming just to keep the discussion focussed. Asterisk is pretty darn stable. I have other Asterisk installs that went well but they were in environments where there were IT people around who were prepared to deal with some Linux administration and I could provide ongoing support for more major things. That is not the case here. Some of those sites have been running for months untouched, some needed some updates and reboots for various issues. I don't think this customer would look very favorably on me having to come in and add patches or have to reboot once a month or whatever. So do it from home. And how often do you really need to upgrade a minimal read only flash based system with no dev tools running from RAM? Does
[asterisk-users] RE: Linksys auto provision
Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Linksys auto provision I have a question about encrypted configs for the Linksys device auto configuration. I am able to do it with xml no problem. However when I generate the text file with the SPC tool then encrypt it with the tool the settings do not take affect. The ATA grabs the correct file but nothing I change is modified when it gets the new config. My guess is that the ATA needs to have the passphrase for the encryption somewhere but none of the fields appear to be labeled passphrase or something intuitive to know where to put it. Any help is appreciated! Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone used vitelity?
Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --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] odd issue with IP tables
I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --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] odd issue with IP tables
-A INPUT # Accept traffic with the ACK flag set -A INPUT -p tcp -m tcp --tcp-flags ACK ACK -j ACCEPT # Allow incoming data that is part of a connection we established -A INPUT -m state --state ESTABLISHED -j ACCEPT # Allow data that is related to existing connections -A INPUT -m state --state RELATED -j ACCEPT -A INPUT -p tcp -m tcp --dport ssh -j ACCEPT -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT -A INPUT -p udp -m udp --dport 5060:5061 -j ACCEPT -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -p udp -m udp --dport 4569 -j ACCEPT And to respond to Alex, The box is only doing Asterisk. 2.8Ghz proc with 1GB of RAM. The iptables is on the server itself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McLeod Sent: Saturday, November 18, 2006 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --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] odd issue with IP tables
Nope the server is for Asterisk only. I have SSH on it for management, FreePBX for configuration, SIP clients and IAX termination. -Original Message- From: Ron McLeod [mailto:[EMAIL PROTECTED] Sent: Saturday, November 18, 2006 8:06 PM To: 'Curt Shaffer'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Do your user agents use some services from the server such as DNS? Ron -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Saturday, November 18, 2006 5:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non- Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables -A INPUT # Accept traffic with the ACK flag set -A INPUT -p tcp -m tcp --tcp-flags ACK ACK -j ACCEPT # Allow incoming data that is part of a connection we established -A INPUT -m state --state ESTABLISHED -j ACCEPT # Allow data that is related to existing connections -A INPUT -m state --state RELATED -j ACCEPT -A INPUT -p tcp -m tcp --dport ssh -j ACCEPT -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT -A INPUT -p udp -m udp --dport 5060:5061 -j ACCEPT -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -p udp -m udp --dport 4569 -j ACCEPT And to respond to Alex, The box is only doing Asterisk. 2.8Ghz proc with 1GB of RAM. The iptables is on the server itself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McLeod Sent: Saturday, November 18, 2006 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --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] voice quality of Aastra 480i CT and cordless
We have had good results mostly from this unit except for one issue that is currently being looked into by Aastra. The issue is if a second call comes in and the cordless answers then puts the call on hold audio drops one way on the handset. Aastra was able to reproduce this and is working on it. From time to time we get reports of bad feedback on the cordless unit but most of the time it is fine. Also the buttons on the cordless are easily mashed with a chubby face :-) Overall we are very pleased with the unit and the ability to have the cordless off of the handset is a great thing. I have not been able to find another unit that has this same feature. Curt _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: Friday, November 17, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice quality of Aastra 480i CT and cordless Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --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] ATA with reliable FAX?
I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what Im looking for: FAX - PTSN - through Asterisk - ATA - Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using in this type of setup with the most luck? Thanks Curt ___ --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] Asterisk VM with Cisco routing
Has anyone out there implemented a system that does call routing via Cisco gear but VM for everyone on the system via Asterisk? What have been your successes and failures or issues? Thanks Curt ___ --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] New Asterisk 1.4 GUI
I was just going to test out the new Asterisk 1.4 GUI. I downloaded it from source make;make install. I added my http.conf and modified manager.conf. I restarted Asterisk and did a make checkconfig and it says everything looks good. But I notice that the port 8088 is not listening when I do a netstat. Am I missing a step here somewhere? Thanks Curt ___ --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] Microsoft will enter VoIP market in earnest
I do not know when they plan on SBS deployment of this. I wouldnt imagine it would not be soon because they just released 2003 R2. The biggest hurdle to this working with Asterisk from what I understand is that it requires SIP over TCP. I havent read the docs fully for 1.4 version of Asterisk is going to support that or not. I am not sure on the storage of the VM either. I would imagine if its not held by Exchange that Exchange will need some kind of rights to the VM server to add/remove/modify/forward VM messages. I have a beta version of it but I just do not have time to install it at the moment. I will be happy to post my results once I do get the time though J Curt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, November 08, 2006 2:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Microsoft will enter VoIP market in earnest Thanks Curt, thats too cool for school, any idea on when this is coming to the MS SBS platform?I use SBS for myself at home and would love that level of functionality included.Does Asterisk therefore handoff voicemail storage etc to Exchange for this level of integration?Cheers,DeanFrom: Curt Shaffer [mailto:cshaffer at gmail.com] Sent: Tuesday, November 07, 2006 11:08 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Microsoft will enter VoIP market inearnestnextyear, says BallmerTake a look at OVA.. http://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv ___ --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] Microsoft will enter VoIP market in earnest nextyear, says Ballmer
Take a look at OVA. mms://wm.microsoft.com/ms/exchange/2007/Phone_Based_User_Experience_With_Outlook_Voice_Access_300k.wmv From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Robar Sent: Tuesday, November 07, 2006 9:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Microsoft will enter VoIP market in earnest nextyear, says Ballmer Unified messaging would be nice. Not just having my VM's e-mailed to me, but to be able to manage them from with Outlook (or any other mail client for that matter) would be nice. I picture it sort of like an IMAP mailbox, and the mail client just has some kind of functionality to recognize that the message is a VM and not a mail message (so it could display length, date/time received, CID, and provide a play button). Just my two cents. Alex On 11/7/06, Dean Collins [EMAIL PROTECTED] wrote: http://www.siliconvalley.com/mld/siliconvalley/business/international/asia/15944981.htm There's not much in the article so only click through if super interested but I'm curious and looking for people's opinions. What application integration would you like to see between MS (either Office or other aspects of the vista/xp OS) and Asterisk. Apart from dial from outlook and number pop I'm kind of curious what other functionality there is to be developed (I'd also like to see drop and drag from outlook into conference calls. What would you like to see in asterisk, if we get some solid responses we'll see about organizing some bounties to get it developed. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] Question on Aastra phones and Astrisk
I'm the friend mentioned here. I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from the PBX to my provider. My issue has a slight twist to it but the same result. For instance his is always where as mine is frequent but not always. After I got to finally see it first hand today, I had to start over from Caller 1 5 times to get it to happen again. Caller 1 calls in and Person A answers. Caller 2 calls in and Person B answers. Person B puts caller 2 on hold and audio drops on Caller 1. So Person A can hear caller 1 but caller 1 cannot hear Person A. This happens more often when Call 1 is on the handset and Call 2 is on the portable or vis a vi, but this is not always the case. It does happen to 1 set only but just less frequent. I have tried carrierinvite=yes and no but this does not change the issue. The phones are behind a router but the external IP of the router is on the same network as the * box. Thanks! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, November 06, 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Aastra phones and Astrisk Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is).We have a connection to and from providers via SIP and IAX.When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me.However, if I park the call, and then pick it up again, the audio is fine. Tonight I tried placing a call on hold using a Sipura/Linksys ATA (that is just hitting 'flash', which basically puts the call on local hold and starts music).The problem did not manifest itself. Has anyone else had this issue? Do you have a fix for it? It is an astrisk issue or an aastra issue? ___ --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] Question on Aastra phones and Astrisk
I wanted to add what we have both seen on traffic captures. You see Caller 1's RTP stream. Call 2 comes in and you see the creation of its RTP stream. After Call 2 is put on hold the RTP stream from Caller 1 disappears without a trace never to return and this is when the one way audio is happening. And I also wanted to add that I am running 1.4.0 firmware for this phone. Thanks again! -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Monday, November 06, 2006 6:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk I'm the friend mentioned here. I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from the PBX to my provider. My issue has a slight twist to it but the same result. For instance his is always where as mine is frequent but not always. After I got to finally see it first hand today, I had to start over from Caller 1 5 times to get it to happen again. Caller 1 calls in and Person A answers. Caller 2 calls in and Person B answers. Person B puts caller 2 on hold and audio drops on Caller 1. So Person A can hear caller 1 but caller 1 cannot hear Person A. This happens more often when Call 1 is on the handset and Call 2 is on the portable or vis a vi, but this is not always the case. It does happen to 1 set only but just less frequent. I have tried carrierinvite=yes and no but this does not change the issue. The phones are behind a router but the external IP of the router is on the same network as the * box. Thanks! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, November 06, 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on Aastra phones and Astrisk Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is).We have a connection to and from providers via SIP and IAX.When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me.However, if I park the call, and then pick it up again, the audio is fine. Tonight I tried placing a call on hold using a Sipura/Linksys ATA (that is just hitting 'flash', which basically puts the call on local hold and starts music).The problem did not manifest itself. Has anyone else had this issue? Do you have a fix for it? It is an astrisk issue or an aastra issue? ___ --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
[asterisk-users] Polycom autoprovision behind a NAT
I am having an issue with doing FTP auto provisioning of Polycom 501s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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 autoprovision behind a NAT
To be honest I dont know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So Im guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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 autoprovision behind a NAT
If you want that is fine. But as I mentioned when I put the phone on the same subnet as the ftp server with no NAT it works like a charm. Is there something in the config that deals with NAT traversal with regards to how it is provisioned? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 8:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT I if you like, I can take a config file(s) and put up over here as a test. Our ftp is working. It might be informative. -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:17:07 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT To be honest I dont know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So Im guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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 autoprovision behind a NAT
Ill try passive and if that doesnt work, Ill email you the configs offline. Thanks for the offer and the help J Curt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 8:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT I'm not sure. We ended up putting in a d-link router to get around the ftp problem. In most of our sites we have netscreen 5gt routers and they work fine. -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:35:27 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT If you want that is fine. But as I mentioned when I put the phone on the same subnet as the ftp server with no NAT it works like a charm. Is there something in the config that deals with NAT traversal with regards to how it is provisioned? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 8:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT I if you like, I can take a config file(s) and put up over here as a test. Our ftp is working. It might be informative. -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 20:17:07 -0600 Subject: RE: [asterisk-users] Polycom autoprovision behind a NAT To be honest I dont know for sure. I am using VSFTPD. I have never needed to set this with setups I have used it before and there is nothing in the config that says passive. So Im guessing that its not. After you asked this I have googled passive FTP and it seems to be on the money as to what is going on so I will try passive and see if that helps. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Monday, November 06, 2006 7:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom autoprovision behind a NAT I can confirm that the linksys routers cause ftp problems. Is your FTP server set to use pasive mode? -rb -Original Message- From: Curt Shaffer [EMAIL PROTECTED] To: ' Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Mon, 6 Nov 2006 19:19:48 -0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT I am having an issue with doing FTP auto provisioning of Polycom 501s when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router. Any ideas? Curt ___ --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] light web user interface
This looks a lot closer to what I need than anything else at this point. Thanks for the link, I'm gonna add start looking at adding functionality to this today! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Rivera Sent: Thursday, November 02, 2006 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] light web user interface Curt Shaffer ([EMAIL PROTECTED]) wrote: Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc. I would also like it open enough that I can add features to it. I'm not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I can't see where you can lock them down to only see information on a particular extension. probably VoiceOne (http://www.voiceone.it/) is wath you need. -- Jonathan Alberto Rivera Gomez Grupo de Usuarios de GNU/Linux - UANL http://linuxuanl.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 ___ --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] light web user interface
Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc I would also like it open enough that I can add features to it. Im not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I cant see where you can lock them down to only see information on a particular extension. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 31, 2006 3:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] light web user interface What attributes are you talking about ? Depending on what they are it may be real simple to set something up. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, October 30, 2006 9:51 PM Subject: [asterisk-users] light web user interface Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___ --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
[asterisk-users] light web user interface
Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___ --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] Open SER or DUNDI
I just wanted to ask a general question to anyone that serves as a service provider on the list out there. Are you using OpenSER and Asterisk for your high availability and redundancy or DUNDI? Anyone have anything to say as to which would be better for a service provider and why? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom provision errors still! Arg!
I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask=255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.0019 24-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=172.16.27.10:ffe0 h=172.16.27.6 g=172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from '172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from '172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/sip.ld' from '172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I dont want to run out of space for the post. Please help! It really cant be this hard. Curt ___ --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 provision errors still! Arg!
Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP= 172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version= 3.2.2.0019 24-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e=172.16.27.10:ffe0 h= 172.16.27.6 g= 172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from '172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from ' 172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/sip.ld' from '172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I don't want to run out of space for the post. Please help! It really can't be this hard. Curt___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Curt Shaffer,MCSA,MCSESecurity+, Network+Certified IP Telephony Sepcialist202-470-6892 (home) 202-470-6893 (Business)309-412-4809 (efax) ___ --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 provision errors still! Arg!
It does boot with the defaults. Is this pointing at a corrupt config? -Original Message- From: Ivan Fetch [mailto:[EMAIL PROTECTED] Sent: Monday, October 23, 2006 6:31 PM To: Curt Shaffer Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom provision errors still! Arg! Hi, I believe he means to use the stock phone1.cfg and mac-address-of-the-phone.cfg files that come with the sip firmware you're running, and see if the phone will load those files. Ivan. On Mon, 23 Oct 2006, Curt Shaffer wrote: Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov-04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.001924-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e= 172.16.27.10:ffe0 h=172.16.27.6 g= 172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from ' 172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from ' 172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/sip.ld' from ' 172.16.27.6' 1023202009|cfg |3|00|Image sip.ld has not changed 1023202009|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr 1 of 1) 1023202009|cfg |3|00|Downloaded application image is identical to current version 1023202009|cfg |3|00|Phone successfully provisioned 1023202041|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 1023202041|app1 |6|00|Uploading boot log, time is MON OCT 23 20:20:42 2006 And it repeats this every time. I can provide the sip.cfg and mac.cfg on request. I don't want to run out of space for the post. Please help! It really can't be this hard. Curt
RE: [asterisk-users] Polycom provision errors still! Arg!
This absolutely helped. I downloaded those config files and copied then and change the name, addressing and such and it worked straight away! Must have been a munged config somehow! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Monday, October 23, 2006 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom provision errors still! Arg! Maybe this might help you. http://www.asterisktutorials.com/showproduct.php?ProductID=12 Cheers, Dean www.Mexuar.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ivan Fetch Sent: Monday, 23 October 2006 7:31 PM To: Curt Shaffer Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom provision errors still! Arg! Hi, I believe he means to use the stock phone1.cfg and mac-address-of-the-phone.cfg files that come with the sip firmware you're running, and see if the phone will load those files. Ivan. On Mon, 23 Oct 2006, Curt Shaffer wrote: Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov- 04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.001924-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e= 172.16.27.10:ffe0 h=172.16.27.6 g= 172.16.27.1 u=pcom pw= tn=CircaIP 1023201827|app1 |3|00|Time has been set from x.x.x.x (x.x.x.x). 1023201827|so |3|00|Link status is Net up Speed 100 full Duplex, PC up Speed 100 full Duplex. 1023201833|cfg |3|00|Beginning to provision phone 1023201833|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/bootrom.ld' from ' 172.16.27.6' 1023201903|cfg |3|00|Image bootrom.ld has not changed 1023201903|copy |3|00|Download of 'bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 1023201903|cfg |3|00|Downloaded bootROM is identical to Current version 3.2.2 1023201903|copy |3|00|'ftp://pcom:[EMAIL PROTECTED]/0004f2027255.cfg' from ' 172.16.27.6' 1023201939|copy |3|00|Download of '0004f2027255.cfg' succeeded on attempt 1 (addr 1 of 1) 1023201939|copy |3|00|' ftp://pcom:[EMAIL PROTECTED]/sip.ld
RE: [asterisk-users] Polycom provision errors still! Arg!
Shit I'll host him for free for that ;) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Monday, October 23, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom provision errors still! Arg! No probs, maybe you should donate $5 to kerry's site to cover hosting fees? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Monday, 23 October 2006 9:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Polycom provision errors still! Arg! This absolutely helped. I downloaded those config files and copied then and change the name, addressing and such and it worked straight away! Must have been a munged config somehow! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Monday, October 23, 2006 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom provision errors still! Arg! Maybe this might help you. http://www.asterisktutorials.com/showproduct.php?ProductID=12 Cheers, Dean www.Mexuar.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ivan Fetch Sent: Monday, 23 October 2006 7:31 PM To: Curt Shaffer Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom provision errors still! Arg! Hi, I believe he means to use the stock phone1.cfg and mac-address-of-the-phone.cfg files that come with the sip firmware you're running, and see if the phone will load those files. Ivan. On Mon, 23 Oct 2006, Curt Shaffer wrote: Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion. Curt On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: What if you just use the default configuration files? On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and point it at the ftp server. It successfully loaded the new firmware and bootrom but will not provision. Every time it gives me Config file error: The error is 0x0 after the page that says Processing Configuration This may take a minute. Here is my ftp log: Mon Oct 23 11:53:18 2006 1 x.x.x.x 339 /home/pcom/0004f2027255.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:53:19 2006 1 x.x.x.x 10240 /home/pcom/sip.ld b _ o r pcom f tp 0 * i Mon Oct 23 11:53:19 2006 1 x.x.x.x 0 /home/pcom/x102\x102.cfg b _ o r pco m ftp 0 * i Mon Oct 23 11:53:27 2006 6 x.x.x.x 121872 /home/pcom/sip.cfg b _ o r pcom ftp 0 * c Mon Oct 23 11:54:07 2006 1 x.x.x.x 9638 /home/pcom/x102/0004f2027255-boot .log b _ i r pcom ftp 0 * c Here is the boot log: |-- Initial log entry -- 1023201556|so |4|00|+++ Note that bootrom log times are in GMT +++ 1023201556|hw |4|00|Initial log entry. 1023201556|wdog |4|00|Initial log entry 1023201556|cfg |4|00|Initial log entry 1023201556|copy |3|00|Initial log entry 1023201556|cdp |4|00|Initial log entry 1023201556|cdp |5|00|CDP is DISABLED. 1023201556|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 1023201556|so |3|00|Platform: Model=SoundPoint IP 501, Assembly=2345-11500-040 Rev=A 1023201556|so |3|00|Platform: Board=2345-11500-040 A 1023201556|so |3|00|Platform: MAC=0004f2027255, IP=172.16.27.10, Subnet Mask= 255.255.255.224 1023201556|so |3|00|Platform: BootBlock=2.5.0 (11500_040) 06-Nov- 04 08:08 1023201556|so |3|00|Application, main: Label=BOOT, Version=3.2.2.001924-Aug-06 18:05 1023201556|so |3|00|Application, main: P/N=3150-11069-322 1023201556|app1 |4|00|Initial log entry. 1023201556|app1 |3|00|DNS resolver servers are 'x.x.x.x' x.x.x.x' 1023201556|app1 |3|00|DNS resolver search domain is '' 1023201556|app1 |3|00|Bootline: eim(0,0)bootHost:flash e= 172.16.27.10:ffe0 h=172.16.27.6 g= 172.16.27.1 u=pcom pw
[asterisk-users] hold drops audio
I have an interesting issue. I have an Aastra 480i CT (the one with the handset and the cordless). Here is the scenario: Caller 1 calls in and the person on the handset answers the call. Caller 2 calls in and the person with the cordless answers the call on the second line (because we call forward on busy to that extension) Caller 2 is put on hold and the audio is lost for Caller 1, never to return. Now if we have this scenario: Caller 1 calls in and the person answers on the handset (or cordless for that matter) Caller 2 calls in and the same person using the same device (whichever was used to answer call 1) answers Caller 2 is put on hold and everything is fine. From this point they can switch between calls and never miss a beat. The issue only appears when the calls are answered on different units. At first I thought this may be an Asterisk issue but now I am thinking it may be an Aastra issue. I plan on calling support when I get on site to troubleshoot but thought I would post and see if anyone else has seen this kind of activity before. *Running Asterisk 1.2.12.1 Thanks Curt ___ --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] WRT54GP2 provisioning
Can anyone point me to a good source for provisioning WRT54GP2 from a central server? Thanks Curt ___ --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] config include issues
Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? [from-internal-custom] exten = 1234,1,Playback(demo-congrats) ; extensions can dial 1234 exten = 1234,2,Hangup() exten = h,1,Hangup() include = NewsClips include = WakeUp [NewsClips] exten = 511,1,Answer exten = 511,2,Wait(1) exten = 511,3,AGI(test.php) exten = 511,4,Hangup [WakeUp] exten = 611,1,Answer exten = 611,2,Playback(demo-congrats) exten = 611,3,Hangup ___ --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] config include issues
[ Context 'from-internal-custom' created by 'pbx_config' ] '1234' = 1. Playback(demo-congrats) [pbx_config] 2. Hangup() [pbx_config] 'h' =1. Hangup() [pbx_config] Include ='NewsClips' [pbx_config] Include ='WakeUp' [pbx_config] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, September 05, 2006 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] config include issues Curt Shaffer wrote: Here is my extensions_custom.conf. The WakeUp context will not work. If I change the context name to say, CRAP, it works like a charm. Can anyone explain this? What does show dialplan from-internal-custom display? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by 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
[asterisk-users] w as pause dialing issue
OK, so I had an issue where I needed to add a w when dialing out my POTS line. But now when the calls go out my VoIP providers the w makes the call fail. I am using freePBX and the only place I found to change this was in the extensions.conf which makes it global. Am I missing something where I can add this while using freePBX? W does not appear to be a valid entry on the trunk prefix or outbound dialing entries. I tried to find a freePBX forum from Google but the only thing that looked promising came up as page cannot be displayed for the past hour. Does anyone have a link to a freePBX forum? I would think this would be a nice feature to add so you can add your pause. I saw where you could add a ticket to the Trac but I would rather discuss it on a list before calling it a needed feature or open ticket. Has anyone experienced this? If so how did you overcome it? Thanks Curt ___ --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] manual mods with GUI in place
This post spurred off of the comment of Michael Collins on the Asterisk with PABX thread. I am going to post the relevant information here: I started w/ AAH, then went back and learned the dialplan apps, scripting, etc. For some guys like me, it's easier to start with a working (if limited) system, and then tinker with it, break it, etc. After breaking a few systems I then went back and did a vanilla install to learn some more. I ended up settling on a compromise: I load Trixbox and then make a bunch of manual mods. I get the best of both worlds - a system that has all of the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd line and also the power and flexibility of hand-editing my .conf files to get exactly what I want out of the dialplan. For those wondering how to get started, I can highly recommend STARTING with Trixbox, but definitely don't STOP with Trixbox. After you play with a pre-installed, working system, go out and get your hands dirty on a plain install. You'll be better off for it in the long run. Having both GUI and cmd line experience will make you a well-rounded Asterisk user. -MC I started w/ AAH, then went back and learned the dialplan apps, scripting, etc. For some guys like me, it's easier to start with a working (if limited) system, and then tinker with it, break it, etc. After breaking a few systems I then went back and did a vanilla install to learn some more. I ended up settling on a compromise: I load Trixbox and then make a bunch of manual mods. I get the best of both worlds - a system that has all of the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd line and also the power and flexibility of hand-editing my .conf files to get exactly what I want out of the dialplan. For those wondering how to get started, I can highly recommend STARTING with Trixbox, but definitely don't STOP with Trixbox. After you play with a pre-installed, working system, go out and get your hands dirty on a plain install. You'll be better off for it in the long run. Having both GUI and cmd line experience will make you a well-rounded Asterisk user. -MC My question to everyone is this..This is where I am at now. I have been using FreePBX for about a year, after moving from [EMAIL PROTECTED] I am starting to need some manual changes and modules. My question is can anyone point me in a direction on how to learn how to create these. I read the ORiley book and thumbed though some of the others, although I plan on reading them all the way through as time permits. I guess my question is where do I add these things. I would still like to use FreePBX because it just saves a ton of coding but I want to add my own things too. Do I put them in the *_additional configs (which appear to be written over by freePBX), the .conf files or the features.conf? Any web links with beginner how tos or more info on this would be appreciated as well! I didnt want to cross post ;) Thanks Curt ___ --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] manual mods with GUI in place
I remember the config edit from [EMAIL PROTECTED] but I do not have it on my freePBX now. I dont mind using vi, Im very comfortable in Linux. Thanks for the answers! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Monday, August 28, 2006 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] manual mods with GUI in place My question to everyone is this..This is where I am at now. I have been using FreePBX for about a year, after moving from [EMAIL PROTECTED] I am starting to need some manual changes and modules. My question is can anyone point me in a direction on how to learn how to create these. I read the ORiley book and thumbed though some of the others, although I plan on reading them all the way through as time permits. I guess my question is where do I add these things. I would still like to use FreePBX because it just saves a ton of coding but I want to add my own things too. Do I put them in the *_additional configs (which appear to be written over by freePBX), the .conf files or the features.conf? Any web links with beginner how tos or more info on this would be appreciated as well! I didnt want to cross post ;) Thanks Curt Curt, First things first I frequently use the FreePBX editor: Logon to your system, then click FreePBX Administration Tools Config Edit You get a nice web-based page where you can bounce around to view and edit all of your config files in /etc and /etc/asterisk Occasionally I am at the Linux cmd line and I use vi, but that is rare. As far as where to start adding your changes, my personal experience is to use the extensions_custom.conf file. This lets me keep my stuff separate from the vanilla install. However, I have made mods to the actual AMP settings to suite my tastes and needs, and for this I did modify extensions.conf. (I keep a backup copy of all of my configs, as Im sure that most of the * users do.) Ive also created completely separate conf files and #included them. Again, this keeps things organized. You can use the #include directive with many of the conf files gurus, please add any known caveats as Ive only used #include for Zapata.conf, extension.conf and sip.conf. As far as how-tos, again I can speak only from experience. There are many how-tos out there, but they are usually pretty specific, so youll probably want to decided WHAT, before you can find a HOW-TO. HtH! -MC ___ --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] hotel teledex integration anyone?
All, I am looking at taking on a project for a hotel that is using Teledex systems. I see that they have a SIP based phone and the information says that there is some CMS server part that appears to be the brains behind the device. My questions are; has anyone out there used this type of system before? Have you integrated it with Asterisk with good success? Any helpful hints when moving into this market? Has any found a comparable CMS (maybe open source) that can be used for this industry? I have been watching the list for these types of posts and I have seen some hotel posts and eagerly read them but have not seen any recommendation of how to best go towards this type of project. Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?
I posted earlier about an application not found error. I have manually pointed the phone at the server but it just does not seem to ever even hit it. I am going to do some network captures here soon after I walk away from this computer for a while. But here is another question which I am not sure if it may be related. After loading the application successfully on other phones I get config error 0x4020 and it just keeps rebooting through this whole process. I have checked my configs and checked them twice against all documentation I could find, and from what I see they are OK. I have posted one here for you all to look at and maybe you can see something I am missing. MAC.cfg (located in /ftproot/ ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=x102/x102.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=x102/ ## X102.cfg (located in /ftproot/x102) ?xml version=1.0 standalone=yes?^M PHONE_CONFIG OVERRIDES reg.1.server.1.expires=60 reg.1.address=102 voIpProt.SIP.outboundProxy.port= log.level.change.cfg=0 _.0x20._log.level.change.sip=0 log.render.level=0 tcpIpApp.sntp.gmtOffset=-21600 tcpIpApp.sntp.address=xxx.xxx.xxx.xxx reg.1.server.1.address=xxx.xxx.xxx.xxx reg.1.auth.password=1234 reg.1.auth.userId=102 voIpProt.server.1.register= reg.1.displayName=Test voIpProt.server.1.address=xxx.xxx.xxx.xxx reg.1.ringType=8/ /PHONE_CONFIG I also have a .cfg file in this directory that has the following: ## .cfg ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone1.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= CONTACTS_DIRECTORY=/ Any help would be appreciated. And I realize this is more of a Polycom question rather than an Asterisk question so if anyone can point me to a good polycom list I would appreciate it as well. Thanks Curt ___ --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] Zap difficulties
That did help. But can you help me understand why this is needed? I did not notice any of the other issues you mentioned but I do notice that it takes an unusually long time to hang up the channel when it is done with the call. It almost seems like the signaling is not right. I was discussing this issue with someone offline and from what I understand, the POTS lines are on loopstart. If that is true why do we use koolstart on the zaptel channel? Just as an experiment I did change the signaling to loopstart but that did not help either. The biggest issue is that I am in an area where just about all of the business are using POTS lines exclusively, and adding a pause to all of these just seems like a hack to me rather than fixing an issue. I'm not saying this is not my misunderstanding, because it may well be, but I am just looking for the exact answer. Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, August 15, 2006 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap difficulties Curt Shaffer wrote: I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Dial(ZAP/g0/w18003569377|120) This will put a .5 second wait before dialing to allow the telco equipment to get ready to receive DTMF. Have you noticed other issues like, even when calling busy numbers, you hear a ringing tone for about 5.5 seconds before you hear a busy tone? That's because you are using the r option to Dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
[asterisk-users] Page Groups
I have a company that I am going to be moving away from a legacy PBX to Asterisk. They use page zones pretty heavy and I would like to keep that functionality. Basically when someone is not at their desk the receptionist pages all of the phones, telling them there is a call. Does anyone out there know of the best phones to do this with and if it is really even possible. I see that intercom is not supported and paging appears to be minimally supported. Thanks Curt ___ --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] IAX unstable with large number of calls?
I was just talking with an unnamed provider and the guy told me that they recommend their users not to use IAX because it is unstable at 50 concurrent calls and unusable at 100 or more calls. Now I have personally worked on an asterisk box that was pushing more than 50 and there were no problems. Anyone else out there have any data either for or against this suggestion? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom upgrade issue
OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___ --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] Zap difficulties
I am having a weird issue with my zap channel (Digium TDM01B). Randomly it appears that the POTS line is not seeing all of the digits passed. We have to dial a 1 and the area code to call most numbers here, and we get the error that we need to dial a 1 and the area code when dialing this number even though we are dialing it. Also when I dial 8xx numbers it never works (same error). I do have all of those set up as allowed and routing properly from the dial plan and I can test that by switching to a VoIP termination and the calls go through without a hitch. I can also dial these numbers fine if I hook a POTS phone directly to the cable that connects to the Digium card. Asterisk looks as if it is passing the digits, (ZAP/g0/18003569377|120|r) for example. Any clues? Thanks Curt ___ --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] Odd IAX stats
Ok, now this may be my lack of understanding on the stats readout of the IAX command but can someone explain the following: I just had two calls going and did an iax2 show channels, the lag for both was 0ms and the jitter was -0001ms. How is that possible? Am I wrong that the lag is the estimated or actual ping time of the remote box? If that is true who on earth has a network so clean that you have ms ping times? Also how does jitter go negative? To add to the oddity, I did iax2 show netstats and the local jitter was -1, the lost was -1 the % was -1 and the remote end never shows any statistics. Is it possible that the remote end can not allow you to see those stats? In addition does Remote even mean remote or does it mean the trip back from remote (along with a post from a user to this list earlier). Thanks Curt ___ --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] Samsung Prostar DCS
I walked into a new potential * install yesterday. They are running a Samsung Prostar DCS. Does anyone have any experience with these out there that you could relay some things to look out for when integrating this until the migration is complete? Or what would be the best way to integrate it while migrating. Thanks Curt ___ --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 VOIP / Mikrotik
And, someone correct me if I am wrong here, you want to make sure RTP is getting quality as well. SIP is setting up, tearing down, and a few other things but RTP is where the conversation is taking place. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Woodoo People .pGa! Sent: Friday, July 28, 2006 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk VOIP / Mikrotik have a 10 mb ethernet connection from my ISP into ether1 on a PC - Mikrotik 2.9.23 installed. ether2 is the rest of my network behind the router. How do I prioritize packets such that VOIP calls ALWAYS get a clean channel through to my Asterisk server, which resides behind that router ? Things sound choppy at best at the moment. not the best, but the easiest way is to check queueing, make a queue dedicated (so channel*(80k if g711||30k if g729)) to voip and max the bandwidth of other=all-voip of course there is an option in mikrotik if you want to dig deeper, to match on udp/sip and give much more priority -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.user s ___ --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 VOIP / Mikrotik
Is it choppy internal or only over the trunk or both? And as far as helping RTP, it should be as simple as adding the ports to your Queue. 1000-2000 by default I believe but you can check your rtp.conf file for the exact. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Smith Sent: Friday, July 28, 2006 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk VOIP / Mikrotik Yep, using SIP for users, IAX for trunks. Can't seem to figure out how to help out the RTP streams though. Once in a while, calls seem clear, but most of the time they're choppy as anything... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Friday, July 28, 2006 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk VOIP / Mikrotik On Jul 28, 2006, at 10:55 AM, Curt Shaffer wrote: And, someone correct me if I am wrong here, you want to make sure RTP is getting quality as well. SIP is setting up, tearing down, and a few other things but RTP is where the conversation is taking place. Yes, if he is using SIP. He didn't mention that. ___ --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 VOIP / Mikrotik
I thought that was not enough zeros but to lazy to look for him ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Friday, July 28, 2006 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk VOIP / Mikrotik On Jul 28, 2006, at 12:12 PM, Curt Shaffer wrote: Is it choppy internal or only over the trunk or both? And as far as helping RTP, it should be as simple as adding the ports to your Queue. 1000-2000 by default I believe but you can check your rtp.conf file for the exact. I think it's 1 to 2 actually... ___ --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
[asterisk-users] Anyone tried vitelity?
I was just wondering if anyone out there has tried vitelity for VoIP service If you did what is your story with how good/bad they are? Thanks! Curt ___ --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] SIP- H323
I have a question. We are going to attempt mixing some SIP and H323 solutions here. The H323 is possibly going to be phased out sooner or later but this is the first step. I have set up an Asterisk server that is also running GnuGK so we have one machine doing both SIP and acting as a Gatekeeper. Both are working in and of themselves but I have not tested the proxy yet. This will probably move to separate boxes once we are out of development. My question is this. If the SIP clients are video capable and the H323 clients are video capable, will that be able to pass the video between? Do the negotiation of the features interoperate? I am quite new to this area, I understand how H323 packets negotiate but I guess I do not know which of these steps have been put into SIP. Has anyone out there tested a scenario like this or similar? If not does anyone have any suggestions on how I could make this possible? Thanks Curt ___ --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] H323 implementation
I have a requirement to set up an Asterisk server that will handle H323. In the end this is used for video conferencing but it will be transitioning other H323 devices to SIP at some point. My question is this: Does anyone know of or have good documentation that explains how this configuration might work or should work. I understand that the implementation of H323 in Asterisk is for a gateway only. I have put GnuGK on the same box to handle the gatekeeper role and they appear to work individually but I have not tested interoperability yet (I will be later this morning). I am supposing that I just point the Asterisk gateway to the gatekeeper (which happens to be on the same box) and it should be able to handle the number mapping. The other problem I have is MCU. I did not have much luck with openMCU yet, so I am in need of that as well. I suppose this turned into a multipoint question, sorry. Has anyone done anything like this out there that was a completely capable unit that will handle (PBX functionality, PSTN connection, and MCU functionality)? Thanks Curt ___ --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: [Asterisk-video] Asterisk as an MCU
Thanks for the information. I guess just as a follow up, is it not possible then to utilize something like MSN messenger or Video capable chat clients that support SIP, like MSN, some sort of jabber or iChat that will allow Asterisk to just pass through the video but handle the voice? I think that would suit our needs for now. Thanks again Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, July 11, 2006 11:05 AM To: Development discussion of video media support in Asterisk Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RE: [Asterisk-video] Asterisk as an MCU Hi Curt, At the moment Asterisk does not perform the functionality you are looking for (there is no single server solution for what you are looking for at the moment). We were looking to sponsor video conferencing development on Asterisk a year ago but put it into the too hard basket. We were then looking to build an application using Adobe Flash media Server but have ceased work on this because of licensing changes which made it uneconomical for less than 100 seats. www.cognation.net/unisona At the moment we use Breeze ASP service to do presentations and Asterisk for Voip (and would use LCS or Jabber for internal messaging but just use MSN messenger). We are doing this with the view that things will change in the next 12 months and will re-look at an all in one service based solution at this time. If I had to buy a video/web presentation server solution at the moment it would be www.wiredred.com Best advice I can offer after spending a lot of time looking at this in the past. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-video- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, 11 July 2006 10:47 AM To: 'Development discussion of video media support in Asterisk' Subject: RE: [Asterisk-video] Asterisk as an MCU Thanks for the clarification. So if I want some functionality of an MCU I could use Asterisk as long as the clients were talking the same (supported) codec? I have never had to build an MCU so I don't know much about them. What we are looking for is video conferencing from workstations through a central system with the ability to dial in from the PSTN and to do IP calls and possibly include some sort of presence features. As far as I can see then Asterisk can fit this bill or am I missing key functionality or performance from not having full MCU capabilities? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey C. Ollie Sent: Tuesday, July 11, 2006 10:41 AM To: Development discussion of video media support in Asterisk Subject: RE: [Asterisk-video] Asterisk as an MCU On Tue, 2006-07-11 at 09:57 -0400, Curt Shaffer wrote: Odd... http://www.voip-info.org/wiki/view/Asterisk+video looks like it does there unless I am missing something. Yes, that page is extremely misleading. Asterisk does not include video codecs. The video support that is mentioned on that page is pass through only. That means that it cannot convert between video formats (which would be required for MCU functionality). Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video ___ --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] Mail loop?
Getting them here too. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk Sent: Tuesday, June 27, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Mail loop? Is anyone else getting messages from the lists.digium.com mail server with errors about a mail loop? I've been getting this for the last few weeks, but I don't have any list software on my server. Any ideas? ___ --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
[Asterisk-Users] SRST type functionality
Has anyone out there figured out how to emulate the Cisco SRST functionality with *? If so would you mind letting me know the best practices for this? Thanks Curt ___ --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] Quality monitoring
Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I dont really know where to start on measuring jitter other than with ICMP pulls and really dont know where to start with doing MOS. Any ideas? Thanks Curt ___ --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] Quality monitoring
It is really just a play on the check_icmp plugin. You could accomplish the same thing by doing the following: $USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1 Where in this example it is an RTA of 80ms or 80% packet loss for a warning and 100ms or 100% packet loss for critical. The perfdata is then passed to perfparse for graphing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, June 22, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quality monitoring Care to share your Nagios plugin? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote: Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I dont really know where to start on measuring jitter other than with ICMP pulls and really dont know where to start with doing MOS. Any ideas? Thanks Curt ___ --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
[Asterisk-Users] RTA, jitter, MOS et al over the internet
I have been in the process of trying to troubleshoot a phone system that is doing IAX trunking to a provider. The average RTA is 75ms with spikes from time to time and jitter from time to time as well. My question is this; How much can one trust this types of samples when going over the internet? I mean who knows who is doing what kind of ICMP rate limiting or dropping ICMP all together? What is a good measurement or troubleshooting step for intermittent bad quality when dealing with links that you have no control over or is that even relevant? Here is our setup: All outbound calls are going out POTS unless that line is congested. All inbound (even from the POTS as it is forwarded directly to an IP DID) are over the IAX trunk. We are not seeing any issues on outbound calls. On inbound, however, we are getting intermittent choppy voice, echo and cutting out. This is heard by the person coming in, i.e when someone is calling they hear these symptoms of the users on the asterisk server. From our side the issue doesnt seem to exist or it is so much less that it is really irrelevant. As I have mentioned I have seen spikes of ping times and times of jitter but this is recognized by tools utilizing ICMP so I dont know how much I can trust them. Thanks for the help! Curt ___ --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] massive screetch and echo from Treo 700w
I am trying to use an IAX softphone (ESCSoftphone) from my Treo 700w. The qualify time is around 173ms. I have only tried setting jitterbuffer=yes in the iax.conf config but the sound is ridiculous. The echo is horrible and there is a screeching in the background on the receive end. Is there anyone out there who a. has any idea to make this usable through their troubleshooting and experience or b. has a suggestion for possibly a better softphone for the Treo? It is running Windows mobile 2005. Thanks Curt ___ --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] Zaptel dialing too fast?
I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am using freePBX 2.1.0, is there a way to do this in there or will this be a manual hack. Here is the tail of the full log when making a call: Jun 15 19:33:51 VERBOSE[25301] logger.c: -- Executing Dial(SIP/103-5595, ZAP/g0/1NXXNXX |120|r) in new stack Jun 15 19:33:51 DEBUG[25301] chan_zap.c: Dialing '1NXXNXX ' Jun 15 19:33:51 DEBUG[25301] chan_zap.c: Deferring dialing... Jun 15 19:33:51 DEBUG[2248] channel.c: Avoiding initial deadlock for 'Zap/1-1' Jun 15 19:33:51 VERBOSE[25301] logger.c: -- Called g0/1NXXNXX Jun 15 19:33:52 DEBUG[25301] chan_zap.c: Exception on 11, channel 1 Jun 15 19:33:52 DEBUG[25301] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) Jun 15 19:33:54 DEBUG[2282] chan_sip.c: Stopping retransmission on '632c53f81b496147556ba1f05f0988e5@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Exception on 11, channel 1 Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Enabled echo cancellation on channel 1 Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Engaged echo training on channel 1 Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Exception on 11, channel 1 Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Echo cancellation already on Jun 15 19:33:56 VERBOSE[25301] logger.c: -- Zap/1-1 answered SIP/103-5595 Jun 15 19:33:56 DEBUG[2282] chan_sip.c: Stopping retransmission on '02730A97-85A3-4FD3-B6EC[EMAIL PROTECTED]' of Response 5200: Match Found Jun 15 19:33:58 DEBUG[2282] chan_sip.c: Auto destroying call ' [EMAIL PROTECTED].xxx.xxx.xxx' Jun 15 19:34:11 NOTICE[25301] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Jun 15 19:34:16 DEBUG[2282] chan_sip.c: Stopping retransmission on '079943eb0215cd9a5d9aec6c4fd96dfb@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:34:16 DEBUG[2300] chan_iax2.c: Peer lastms 71, historicms 71, maxms 2000 Jun 15 19:34:16 DEBUG[2282] chan_sip.c: Stopping retransmission on '3e4904875c7aaa750cacd89a7d94e891@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:34:17 DEBUG[2282] chan_sip.c: Stopping retransmission on '6bb8c02a2549ac2104f44f9145e5fba9@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on '5069f5f5785e9c6770ff6f815117646d@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on '1ba2bc38733dfd693cde414e200a9546@ xxx.xxx.xxx.xxx' of Request 102: Match Found Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on ' 0ccc55236435810960f805fd7124feb[EMAIL PROTECTED]' of Request 102: Match Found Jun 15 19:34:23 DEBUG[25301] channel.c: Didn't get a frame from channel: SIP/103-5595 Jun 15 19:34:23 DEBUG[25301] channel.c: Bridge stops bridging channels SIP/103-5595 and Zap/1-1 Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Hangup: channel: 1 index = 0, normal = 11, callwait = -1, thirdcall = -1 Jun 15 19:34:23 DEBUG[25301] chan_zap.c: disabled echo cancellation on channel 1 Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Updated conferencing on 1, with 0 conference users Jun 15 19:34:23 VERBOSE[25301] logger.c: -- Hungup 'Zap/1-1' Jun 15 19:34:23 DEBUG[25301] app_dial.c: Exiting with DIALSTATUS=ANSWER. Thanks, Curt ___ --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] rollover simulation
I am trying to perform a rollover when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I dont have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number (which points to another successfully registered line on the same phone) and number of rings 1 (although I have tried 2 and 3). This setting is on the line but I have tried global as well also. Any clues? Thanks Curt ___ --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] No CID on ZAP
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbound are providing CID with no issue. Here is the output from the log when a call is coming in: -- Starting simple switch on 'Zap/1-1' -- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack -- Executing Set(Zap/1-1, DID=s) in new stack -- Executing NoOp(Zap/1-1, DID is now s) in new stack -- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack -- Goto (from-zaptel,s,7) -- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack -- Executing Set(Zap/1-1, CHAN=1-1) in new stack -- Executing Set(Zap/1-1, CHAN=1) in new stack -- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack -- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new stack -- Executing Goto(Zap/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-local|200|1) in new stack -- Goto (ext-local,200,1) -- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing GotoIf(Zap/1-1, 0?report) in new stack -- Executing GotoIf(Zap/1-1, 0?start) in new stack -- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack -- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack -- Executing Set(Zap/1-1, AMPUSER=) in new stack -- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Zap/1-1, 1?report) in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Set(Zap/1-1, VMBOX=200) in new stack -- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack -- Executing Set(Zap/1-1, CFUEXT=) in new stack -- Executing Set(Zap/1-1, RT=25) in new stack -- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060609-095557|1149864957.408) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060609-095557|1149864957.408: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack -- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- dialparties.agi: Checking CW and CFB status for extension 200 -- dialparties.agi: DbSet CALLTRACE/200 to unknown -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack Any help would be appreciated. Thanks Curt ___ --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] Re: No CID on ZAP
Someone mentioned that it may need a pause because the CID is sent between ring 1 and ring 2. So now if this is the case, I am trying to enter this pause. I am using freePBX 2.1.0 for configuration. So I went into extensions.conf
to find the from-zaptel and this is what I tried to add:
[from-zaptel]exten = _X.,1,Wait(2)exten = _X.,n,Set(DID=${EXTEN})exten = _X.,n,Goto(s,1)exten = s,1,NoOp(Entering from-zaptel with DID == ${DID}); If ($did == ) { $did = s; }
The wait statement seems to be ignored. Can anyone out there point me to the right direction to get this to function properly?
On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote:
I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this
channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbound are providing CID with no issue.
Here is the output from the log when a call is coming in:
-- Starting simple switch on 'Zap/1-1'
-- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack
-- Executing Set(Zap/1-1, DID=s) in new stack
-- Executing NoOp(Zap/1-1, DID is now s) in new stack
-- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack
-- Goto (from-zaptel,s,7)
-- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack
-- Executing Set(Zap/1-1, CHAN=1-1) in new stack
-- Executing Set(Zap/1-1, CHAN=1) in new stack
-- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack
-- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new stack
-- Executing Goto(Zap/1-1, ext-did|s|1) in new stack
-- Goto (ext-did,s,1)
-- Executing Set(Zap/1-1, FROM_DID=s) in new stack
-- Executing Goto(Zap/1-1, ext-local|200|1) in new stack
-- Goto (ext-local,200,1)
-- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack
-- Executing Macro(Zap/1-1, user-callerid) in new stack
-- Executing GotoIf(Zap/1-1, 0?report) in new stack
-- Executing GotoIf(Zap/1-1, 0?start) in new stack
-- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack
-- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack
-- Executing Set(Zap/1-1, AMPUSER=) in new stack
-- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack
-- Executing GotoIf(Zap/1-1, 1?report) in new stack
-- Goto (macro-user-callerid,s,9)
-- Executing NoOp(Zap/1-1, Using CallerID ) in new stack
-- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack
-- Executing Set(Zap/1-1, VMBOX=200) in new stack
-- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack
-- Executing Set(Zap/1-1, CFUEXT=) in new stack
-- Executing Set(Zap/1-1, RT=25) in new stack
-- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack
-- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Zap/1-1, recordingcheck|20060609-095557|1149864957.408) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060609-095557|1149864957.408: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(Zap/1-1, No recording needed) in new stack
-- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack
-- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack
-- Executing AGI(Zap/1-1, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- dialparties.agi: Checking CW and CFB status for extension 200
-- dialparties.agi: DbSet CALLTRACE/200 to unknown
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack
Any help would be appreciated.
Thanks
Curt-- Curt Shaffer,MCSA,MCSE
Security+, Network+Certified IP Telephony Sepcialist202-470-6892 (home)1-309-412-4809 (efax)202-470-6893 (Business)570-207-1822 (fax)
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RE: [Asterisk-Users] No CID on ZAP
[channels] language=en #include zapata_additional.conf context=from-zaptel signalling=fxs_ks faxdetect=incoming usecallerid=asreceived echocancel=yes callprogress=no busydetect=no echocancelwhenbridged=no echotraining=800 group=0 channel=1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Friday, June 09, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No CID on ZAP Thanks for sharing that info. How about sharing your zapata.conf configuration so that someone can look at it and maybe see if there is a problem. I'm guessing you want help with this. On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote: I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbound are providing CID with no issue. Here is the output from the log when a call is coming in: -- Starting simple switch on 'Zap/1-1' -- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack -- Executing Set(Zap/1-1, DID=s) in new stack -- Executing NoOp(Zap/1-1, DID is now s) in new stack -- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack -- Goto (from-zaptel,s,7) -- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack -- Executing Set(Zap/1-1, CHAN=1-1) in new stack -- Executing Set(Zap/1-1, CHAN=1) in new stack -- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack -- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new stack -- Executing Goto(Zap/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-local|200|1) in new stack -- Goto (ext-local,200,1) -- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing GotoIf(Zap/1-1, 0?report) in new stack -- Executing GotoIf(Zap/1-1, 0?start) in new stack -- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack -- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack -- Executing Set(Zap/1-1, AMPUSER=) in new stack -- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Zap/1-1, 1?report) in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Set(Zap/1-1, VMBOX=200) in new stack -- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack -- Executing Set(Zap/1-1, CFUEXT=) in new stack -- Executing Set(Zap/1-1, RT=25) in new stack -- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060609-095557|1149864957.408) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060609-095557|1149864957.408: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack -- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- dialparties.agi: Checking CW and CFB status for extension 200 -- dialparties.agi: DbSet CALLTRACE/200 to unknown -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack Any help would be appreciated. Thanks Curt ___ --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 -- Tom Vile Baldwin Technology
RE: [Asterisk-Users] No CID on ZAP
That did it! Thanks a million! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Friday, June 09, 2006 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No CID on ZAP Mine has usecallerid=yes and caller id works. Not sure if that's the problem or not. On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote: [channels] language=en #include zapata_additional.conf context=from-zaptel signalling=fxs_ks faxdetect=incoming usecallerid=asreceived echocancel=yes callprogress=no busydetect=no echocancelwhenbridged=no echotraining=800 group=0 channel=1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Tom Vile Sent: Friday, June 09, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No CID on ZAP Thanks for sharing that info. How about sharing your zapata.conf configuration so that someone can look at it and maybe see if there is a problem. I'm guessing you want help with this. On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote: I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbound are providing CID with no issue. Here is the output from the log when a call is coming in: -- Starting simple switch on 'Zap/1-1' -- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack -- Executing Set(Zap/1-1, DID=s) in new stack -- Executing NoOp(Zap/1-1, DID is now s) in new stack -- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack -- Goto (from-zaptel,s,7) -- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack -- Executing Set(Zap/1-1, CHAN=1-1) in new stack -- Executing Set(Zap/1-1, CHAN=1) in new stack -- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack -- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) in new stack -- Executing Goto(Zap/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-local|200|1) in new stack -- Goto (ext-local,200,1) -- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing GotoIf(Zap/1-1, 0?report) in new stack -- Executing GotoIf(Zap/1-1, 0?start) in new stack -- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack -- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack -- Executing Set(Zap/1-1, AMPUSER=) in new stack -- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Zap/1-1, 1?report) in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Set(Zap/1-1, VMBOX=200) in new stack -- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack -- Executing Set(Zap/1-1, CFUEXT=) in new stack -- Executing Set(Zap/1-1, RT=25) in new stack -- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060609-095557|1149864957.408) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060609-095557|1149864957.408: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack -- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi --dialparties.agi: priority is 1dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is'none' --dialparties.agi: Added extension 200 to extension map --dialparties.agi: Extension 200 cf is disabled --dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 --dialparties.agi: Checking CW and CFB status for extension 200 --dialparties.agi : DbSet CALLTRACE/200 to unknown -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack Any help would be appreciated. Thanks Curt
RE: [Asterisk-Users] Vonage and FXO
I used a scenario like this before but I always ran into intermittent echo issues that were just not worth the hassle for me so I switched to a sole IP origination and termination service. Just my personal experience! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, June 06, 2006 12:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and FXO Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --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] Vonage and FXO
Sorry, I guess that would help! I was using an X100P so I am sure that was a large part of the problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Tuesday, June 06, 2006 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vonage and FXO It would be helpful if responders would tell us what FXO hardware they are using and which vonage ATA device it connects to. Padmanaban Balasubramaniam wrote: I am using it with my [EMAIL PROTECTED] setup. I did not face any issues with echo, but once in a while, the trunk does NOT get disconnected even after the call has been completed. So I had to manually plug the phone cable out from FXO and plug it back again. But I think that's something to do with my version of FXO drivers. Otherwise it works for me. Paddu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Tuesday, June 06, 2006 10:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Vonage and FXO I used a scenario like this before but I always ran into intermittent echo issues that were just not worth the hassle for me so I switched to a sole IP origination and termination service. Just my personal experience! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mustardman29 Sent: Tuesday, June 06, 2006 12:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and FXO Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial fxo port 5 (Vonage ATA) first and if it's used then use fxo ports 1-4. Is this easy to do in FreePBX? I know I can get a Vonage softphone account and not use an ATA/FXO port. I want to know if I can do it with an ATA/FXO. ___ --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 ___ --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] Configuring Polycom 501 IP phones via the console
I too had the same problems. If you find out the best way for this let me know! Thanks Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Sunday, June 04, 2006 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Configuring Polycom 501 IP phones via the console Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] TDM
Found the issue with the help of Digium. The system we were using was to be only IP once upon a time so I did not compile zaptel initially. I did before I installed the card but I needed to recompile asterisk so it added the zaptel support. I hate it when it's something like that ;P Thanks for all of your suggestions! Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Melcon Moraes Sent: Sunday, May 28, 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re[2]: [Asterisk-Users] TDM What if you try Zap instead of ZAP for channel name? []'s MM -Original Message- From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Sun, 28 May 2006 13:33:46 -0400 Delivered: Sun, 28 May 2006 14:28:38 Subject:[Asterisk-Users] TDM It looks OK. Try editing extensions.conf and add an extension in a context that will included when you dial. Try something like this exten = 123,1,Dial(ZAP/g0/1NXXNXX) The open the console and dial 123. This will bypass any funky dialplan issues with FreePBX. If it works, then obviously something is not right in FreePBX. If it doesnt' then that indicates your configuration files need tweaking. Thanks, Steve Curt Shaffer wrote: Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing
[Asterisk-Users] Polycom 501
Does anyone out there have a sample config they can share for the Polycom 501? Is it possible to do sub configs like you can with the Aastra 9133i? It could be just me but the boot configs seem a bit cryptic compared to the aastra. Also do any of you have any comparisons between these and the Aastra 9133i? Thanks Curt ___ --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] TDM
Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps
RE: [Asterisk-Users] TDM
Here is the output from a dial when starting asterisk with -v. The 1NXXNXX is actually the number not those characters FYI. Thanks -- Executing Macro(SIP/103-a555, dialout-trunk|1|1NXXNXX||) in new stack -- Executing GotoIf(SIP/103-a555, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/103-a555, user-callerid) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing GotoIf(SIP/103-a555, 0?start) in new stack -- Executing Set(SIP/103-a555, REALCALLERIDNUM=103) in new stack -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, AMPUSER=103) in new stack -- Executing Set(SIP/103-a555, AMPUSERCIDNAME=103) in new stack -- Executing GotoIf(SIP/103-a555, 0?report) in new stack -- Executing Set(SIP/103-a555, CALLERID(all)=103 103) in new stack -- Executing NoOp(SIP/103-a555, Using CallerID 103 103) in new stack -- Executing Macro(SIP/103-a555, record-enable|103|OUT) in new stack -- Executing GotoIf(SIP/103-a555, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/103-a555, recordingcheck|20060528-110627|1148832387.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060528-110627|1148832387.1: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/103-a555, No recording needed) in new stack -- Executing Macro(SIP/103-a555, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/103-a555, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/103-a555, REALCALLERIDNUM is 103) in new stack -- Executing Set(SIP/103-a555, USEROUTCID=) in new stack -- Executing Set(SIP/103-a555, EMERGENCYCID=) in new stack -- Executing Set(SIP/103-a555, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/103-a555, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(SIP/103-a555, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf(SIP/103-a555, 1?report) in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp(SIP/103-a555, CallerID set to 103 103) in new stack -- Executing Set(SIP/103-a555, GROUP()=OUT_1) in new stack -- Executing GotoIf(SIP/103-a555, 0?108) in new stack -- Executing Set(SIP/103-a555, DIAL_NUMBER=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/103-a555, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/103-a555, OUTNUM=1NXXNXX) in new stack -- Executing Set(SIP/103-a555, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/103-a555, 0?16) in new stack -- Executing Dial(SIP/103-a555, ZAP/g0/1NXXNXX|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(SIP/103-a555, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(SIP/103-a555, Dial failed due to CHANUNAVAIL) in new stack -- Executing Macro(SIP/103-a555, outisbusy|) in new stack -- Executing Playback(SIP/103-a555, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback(SIP/103-a555, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' in macro 'outisbusy' == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/103-a555' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, May 28, 2006 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Connect to the Asterisk console with verbose turned on and try to dial. Post that output. Curt Shaffer wrote: This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps
[Asterisk-Users] TDM
The TDM01B is 4 port capable but has only 1 FXO module. Im running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include zapata_additional.conf context=from-zaptel signalling=fxs_ks faxdetect=incoming usecallerid=asreceived echocancel=yes callprogress=no busydetect=no echocancelwhenbridged=no echotraining=800 group=0 channel=1 When I dial in Asterisk does not even show an initiation of the call. When I dial out on that trunk I get all circuits busy. Ztcfg vvv shows the following ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Any help would be appreciated. Curt ___ --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] TDM
This is not [EMAIL PROTECTED] it is asterisk with FreePBX only. Yes the phone line is connected to the right port. No luck. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Saturday, May 27, 2006 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM Steve Totaro wrote: Is your machine seeing the card? /var/log/messages? Are you loading the zaptel drivers? modprobe zaptel, modprobe wctdm? Would he get the ztcfg message if it were not? Is the phone line plugged into the correct jack? With only one module installed, the other three jacks lead to nowhere. Also this seems to be [EMAIL PROTECTED] from the references, so perhaps there is a context issue that the configuration files address. AAH can really lead one down the garden path! John Novack Curt Shaffer wrote: The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include zapata_additional.conf context=from-zaptel signalling=fxs_ks faxdetect=incoming usecallerid=asreceived echocancel=yes callprogress=no busydetect=no echocancelwhenbridged=no echotraining=800 group=0 channel=1 When I dial in Asterisk does not even show an initiation of the call. When I dial out on that trunk I get all circuits busy. Ztcfg -vvv shows the following ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Any help would be appreciated. Curt ___ --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
[Asterisk-Users] SIP Video software
All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they have used successfully with Asterisk. Thanks Curt smime.p7s Description: S/MIME cryptographic signature ___ --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] Video SIP Softset
Sorry if this shows twice but it appears my first message was quarantined because of my digital signature. All, I have been tasked with setting up video conferencing utilizing asterisk. One of the requirements is a softset that has video capabilities. Eyebeam looks promising but I was just wondering if anyone out there knows of any freeware with comparable features of Eyebeam that they have used successfully with Asterisk. Thanks Curt ___ --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] Company List
I have not but if you find one, please pass it on because I have the same requirement. Curt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, April 12, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Company List The question was raised by a CFO who is looking at Asterisk if there is a list of companies using Asterisk. I have not found one yet, has anyone seen anything like this I can give him. -- Bruce Nortex Networks ___ --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] Company List
I disagree a bit. A lot of companies publish their customer list for reasons of advertisement. If I have a client that is joe blow fortune 500 company, I'm gonna publish that for my credibility. I think that is what we are looking for (I think I can safely speak for both of us on this). Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, April 12, 2006 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Company List I would doubt that anyone is going to share their customer list for obvious reasons. I'd have to guess that in access of 80% of the production implementations are sold by resellers (of various sizes), and maybe 20% are actual in-house implementations by those that frequent this list. The 80% is probably what you'd be interested in, but not likely to be published anywhere. Curt Shaffer wrote: I have not but if you find one, please pass it on because I have the same requirement. Curt *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Wednesday, April 12, 2006 3:51 PM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] Company List The question was raised by a CFO who is looking at Asterisk if there is a list of companies using Asterisk. I have not found one yet, has anyone seen anything like this I can give him. -- Bruce Nortex Networks ___ --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
[Asterisk-Users] sixtel
Just wondering everyones experience with Six Tel (http://www.iax.cc/show.php?go=local)? They seem to have some really decent prices but I have heard some buyer beware comments elsewhere. Thanks Curt ___ --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 with Vonage
I have this working. I have Asterisk connecting to my Vonage Linksys device via Digium Wildcard X100P. No magic needed ;) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: Wednesday, March 29, 2006 9:25 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk with Vonage Plus see this: http://www.voip-info.org/wiki/view/Asterisk+and+Vonage Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, March 29, 2006 8:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Vonage On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! I'm not a Vonage customer, but I did spot this: http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credenti als.asp Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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 EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. EMAIL PRIVELGED CONFIDENTIAL CLIENT COMMUNICATION.DOCDKH ___ --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
[Asterisk-Users] eyeBeam v1.1
Has anyone out there used eyeBeam v1.1 with Asterisk? If so what kind of results do you have? Thanks Curt smime.p7s Description: S/MIME cryptographic signature ___ --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] Receptionist Phones (was 3Com Phones)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Monday, March 27, 2006 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist Phones (was 3Com Phones) Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to, but would like to be able to cover all the phones) and see who is on the phone already. I would like to avoid a software solution as those tend to be confusing and hard for non-computer savvy people to deal with. I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker ___ --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 Have you looked that the flash operator panel? http://www.asternic.org/demo.html I know you mentioned not wanting a software solution because of confusion but I think that would be pretty easy to understand. Curt ___ --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] FXS channel banks
Title: Message As of now we are probably looking in the 36 range. We would like to utilize this as a first step to migrating to a VoIP system. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 25, 2006 2:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] FXS channel banks How many phones lines ? -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Curt Shaffer Envoyé: vendredi 24 mars 2006 03:17 À: asterisk-users@lists.digium.com Objet: [Asterisk-Users] FXS channel banks Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt smime.p7s Description: S/MIME cryptographic signature ___ --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] 3Com Phones
I would not recommend the 3Com phones. I know to get most of them to even work on 3Com systems you need to purchase licenses. For the prices you want to pay you would definitely be better off going with something else. The list price for the 3101 is $155 The list price for the 3102 is $240 The list price for the 3103 is $365 The list price for the 3105 is $255 Phone licensing is list price of about $135/year Of course a partner could probably give you a little better of a deal depending on your relationship with them. I am freshly out of a 3Com only world so I cannot point you in the exact direction but I am sure you can get comparable phones from places like Polycom and others. Maybe these prices can give others on the list an idea of what you are looking at spending. I would stay away from anything 3Com if you want a compatible, fully functional system (Pretty scary statement from being certified in 3Com IP telephony ;)) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stoffell Sent: Saturday, March 25, 2006 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3Com Phones On 3/25/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: We are looking at installing a VoIP system with Asterisk and are currently looking at the line of 3Com phones. Has anybody had success with using the following phones? We need to buy a lot and we don't want to end up with phones that don't work properly with asterisk. I didn't even know 3Com had VoIP phones, I'm also curious on these.. How many phones do you need and what is your budget and features wishlist? cheers ___ --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 smime.p7s Description: S/MIME cryptographic signature ___ --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] FXS channel banks
Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt smime.p7s Description: S/MIME cryptographic signature ___ --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] Failed installing zaptel
I had the same kind of issue myself. The kernel was upgrading to 2.6.9-34 from 2.6.9-22 but for some reason it did not appear that way to the compiler. I reinstalled Cent OS 4.2 and updated everything except for the kernel and did a wget for the 2.6.9-22 source from the mirror and it worked like a charm! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Failed installing zaptel Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock'
[Asterisk-Users] Re: Zap not installing
This issue has been solved. What I found was that [EMAIL PROTECTED] was running a newer version of udev. Once I installed the newer version it came right up. The version udev-039-10.10.EL4.3 works like a charm! Hope that helps someone out there. Curt On 3/8/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have decided to move on from [EMAIL PROTECTED] and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of O'Reilly's Asterisk the future of technology and begun. I downloaded the zaptel -1.2.4.tar.gz , libpri-1.0.9, and asterisk-1.2.5. I started compiling the zaptel (make make install make clean) when I try to start zaptel - /etc/init.d/ zaptel start I get the following error: Loading zaptel framework: FATAL: Module zaptel not found Unable to open /dev/zap/ctl: No such file or directory Below are the only things I have declared in my /etc/zaptel.conf ks=1 loadzone=us defaultzone=us fxoks=1 ( I have tried fxsks=1 as well, because the book had a section that read the following): ...a physical FXO port will be defined in configuration with FXS signaling..an FXO card connects to a central office(CO), which means it will need to behave like a station that use FXS signaling I tried this both in /etc/udev/rules.d/50-udev.rules and /etc/udev/rules.d/zaptel.rules (rebooting after each change) Zaptel devices KERNEL=zapctl, NAME=zap/ctl KERNEL=zaptimer, NAME=zap/timer KERNEL=zapchannel, NAME=zap/channel KERNEL=zappseudo, NAME=zap/pseudo KERNEL=zap[0-9]*, NAME=zap/%n When I run ztcfg I get the following error: line 0: Unable to open master device '/dev/zap/ctl' When I run zttool I get the following error: Unable to open /dev/zap/ctl: No such file or directory I have started from scratch multiple times and I get the same result. I get no errors when compiling and the card can be removed and put back in the old system and work properly. Also Linux does notice the device when I install and boot into the OS. Any help would be appreciated. Curt -- Curt Shaffer, Network+,MCP, MCSA202-558-2408 (home)1-309-412-4809 (efax) ___ --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] Zap not installing
I have decided to move on from [EMAIL PROTECTED] and start compiling asterisk myself now. I got a dedicated box and put my X100P in it. I installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of OReillys Asterisk the future of technology and begun. I downloaded the zaptel-1.2.4.tar.gz, libpri-1.0.9, and asterisk-1.2.5. I started compiling the zaptel (make make install make clean) when I try to start zaptel - /etc/init.d/zaptel start I get the following error: Loading zaptel framework: FATAL: Module zaptel not found Unable to open /dev/zap/ctl: No such file or directory Below are the only things I have declared in my /etc/zaptel.conf ks=1 loadzone=us defaultzone=us fxoks=1 ( I have tried fxsks=1 as well, because the book had a section that read the following): ...a physical FXO port will be defined in configuration with FXS signaling..an FXO card connects to a central office(CO), which means it will need to behave like a station that use FXS signaling I tried this both in /etc/udev/rules.d/50-udev.rules and /etc/udev/rules.d/zaptel.rules (rebooting after each change) Zaptel devices KERNEL=zapctl, NAME=zap/ctl KERNEL=zaptimer, NAME=zap/timer KERNEL=zapchannel, NAME=zap/channel KERNEL=zappseudo, NAME=zap/pseudo KERNEL=zap[0-9]*, NAME=zap/%n When I run ztcfg I get the following error: line 0: Unable to open master device '/dev/zap/ctl' When I run zttool I get the following error: Unable to open /dev/zap/ctl: No such file or directory I have started from scratch multiple times and I get the same result. I get no errors when compiling and the card can be removed and put back in the old system and work properly. Also Linux does notice the device when I install and boot into the OS. Any help would be appreciated. Curt ___ --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
