Re: [asterisk-users] ztdummy and MOH
On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I am using autoload and I have rebooted the server. I have tried using different files and a different location. This is getting very frustrating. I wish I knew what the problem was. Not that it will help me, because I'm pretty much clueless from here... How are you calling music on hold? Is this just by putting someone on hold, or do you have an extension defined with music on hold? Someone else may be able to pick up on something by knowing that. In the forums, there's a mention of music on hold having problems when used with the dial command (instead of ringing, use music on hold). I know there is an issue using that in connection with chan_sccp and Cisco phones. I've run out of ideas. Hopefully someone in the morning that's had this issue will see this. It kinda rules out any bad files or conversion problems if you've used the files that came with asterisk. I hate being stumped, but I'm stumped. ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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] Counting callers
Matt wrote: Do you mean queue? If so, yes this is a very easy thing to do and is document on the voip-info.org http://voip-info.org wiki under the queues section. Thank you and excuse me I'am a totally newbie in VoIP and tel! I solved my problem with queue. On 3/26/07, * Suity Zsolt* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Can I count and say back to caller how many calls waiting on current extension? -- Suich ___ --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] SRTP vs ZRTP in Asterisk
27 mar 2007 kl. 10.48 skrev Tim Panton: On 26 Mar 2007, at 22:32, Michael Graves wrote: Hi All, I've been reading about Phil Zimmermann's ZRTP encryption scheme for SIP clients. This seems attactive but I don't use soft phones. I'm guessing that we'd need ZRTP support in Asterisk in order to use it to secure calls from hard phones. There seem to be issues with SRTP key exhange between various devices. So much so that the IETF is working on a standardization project. ZRTP, which is one of the proposals before the IETF, overcomes this. Since Zimmermann has open sourced the protocol I would hope that it could be implemented in Asterisk without too much trouble. Does the current work on SRTP extend into ZRTP? At Etel I heard Phil Zimmermann say that he had a working implementation of ZRTP for asterisk in the lab. We are working on the legal issues as well as issues on how to integrate this properly, both from a code standpoint and a security standpoint. /O ___ --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] just on my LAN
hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 Zaptel 1.2.16 Libpri 1.2.4 Addons 1.2.5 Sounds 1.2.1 thanks ___ --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 MeetMe bahaviour with MoH ...
Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten = 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH stops. Log shows this: -- Starting simple switch on 'Zap/1-1' -- Executing MeetMe(Zap/1-1, 400|iM) in new stack -- Playing 'conf-getpin' (language 'en') -- Recording -- Playing 'vm-rec-name' (language 'en') -- Stopped music on hold on SIP/101-0816b2a8 -- Playing 'beep' (language 'en') So it looks like it's possibly a deliberate action - but the effect on the 1st person is that all goes quiet - and so they think it's broken, so they hang-up ... (or have a tendency to!) So is this to be expected, just an oversight in the code, or am I doing something obviously silly... (Asterisk 1.2.16 FWIW) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I generate random SIP traffic?
Hello, I would like to generate a peer-to-peer or a server/client SIP traffic between two or more Openwrt access point, to make some statistics about QoS. I tried some SIP traffic generators for OpenWrt, but I didn't find nothing of satisfactory. Now I wonder if asterisk can help me generating random SIP traffic. I'm googling since yesterday without results. Can you help me plz? Thanks and sorry for the disturb. ___ --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 MeetMe bahaviour with MoH ...
This is the expected behavior -- if the second person comes in and you have name announcements, then the first person will hear that and should have the sense to know not to hang up. You can have everybody hear music till a certain person comes in, if you want. on Wednesday 03/28/2007 Gordon Henderson([EMAIL PROTECTED]) wrote Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten = 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH stops. Log shows this: -- Starting simple switch on 'Zap/1-1' -- Executing MeetMe(Zap/1-1, 400|iM) in new stack -- Playing 'conf-getpin' (language 'en') -- Recording -- Playing 'vm-rec-name' (language 'en') -- Stopped music on hold on SIP/101-0816b2a8 -- Playing 'beep' (language 'en') So it looks like it's possibly a deliberate action - but the effect on the 1st person is that all goes quiet - and so they think it's broken, so they hang-up ... (or have a tendency to!) So is this to be expected, just an oversight in the code, or am I doing something obviously silly... (Asterisk 1.2.16 FWIW) Cheers, Gordon ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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
[asterisk-users] REG : H.323 Configurations with Asterisk
Hi , I am new to Asterisk community. I have some queries. Please guide me on the following : 1)I want to configure H.323 softphones, How do I do that ? I am using the Asterisk windows versio 0.60.There is no chan_h.323.so file .Also there are no help files or documents for configuring h.323 softphones. Can someone guide me inthis regard ? 2)And I want to know how the supplementary features for the H.323 like Transfer,hold and forward are handled in Asterisk. Is that part of the OpenH323 library or the Asterisk takes care of them. Regards, Anisha ___ --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] UK BT PRI
hi, i don't know if this will work or not but i've a friend working in siemens that tell me to work with a PRI software tracer like what he has, i still looking for a one working on linux asterisk, using the tracer log , you can find how many digits are used :) i don't know i wish this help kinf regards ___ --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 MeetMe bahaviour with MoH ...
On Wed, 28 Mar 2007, John covici wrote: This is the expected behavior -- if the second person comes in and you have name announcements, then the first person will hear that and should have the sense to know not to hang up. You can have everybody hear music till a certain person comes in, if you want. Intersting - but what I'm observing is that MoH to No. 1 stops at the point when No. 2 is asked to speak their name - if they take a long time, (eg. pressing 3 to review, etc.) the No.1 is left in silence until No. 2 accepts their name recording... After that, it's fine as others join as theires no MoH, however it's just the pregnant pause for No. 1 after they've already been listening to MoH when waiting for No. 2 that I was curious about... Cheers, Gordon on Wednesday 03/28/2007 Gordon Henderson([EMAIL PROTECTED]) wrote Hi, I've just observed something a bit odd - I'm wondering if this is the expected behaviour, a bug/feature, or something I'm doing stupid! 1st person gets into MeetMe. Nothing fancy, just: exten = 987,1,MeetMe(400,iM) They enter the passcode and their name, then listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH stops. Log shows this: -- Starting simple switch on 'Zap/1-1' -- Executing MeetMe(Zap/1-1, 400|iM) in new stack -- Playing 'conf-getpin' (language 'en') -- Recording -- Playing 'vm-rec-name' (language 'en') -- Stopped music on hold on SIP/101-0816b2a8 -- Playing 'beep' (language 'en') So it looks like it's possibly a deliberate action - but the effect on the 1st person is that all goes quiet - and so they think it's broken, so they hang-up ... (or have a tendency to!) So is this to be expected, just an oversight in the code, or am I doing something obviously silly... (Asterisk 1.2.16 FWIW) Cheers, Gordon ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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 ___ --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: How is this feature called ?
Olivier wrote: Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Hi Oliver, is this some new feature that you have invented and you need to come up with some name for it? -- Tomislav Parcina [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
[asterisk-users] Voicemailmain not changing password?
hi all, i am using voicemailmain application in ast 1.4.2. Its not changing my password in the change password menu. i have no idea why. my voicemail configuration is: 25= 52,sipura i always have to enter 52 for password even if i have changed it previously. can anyone tell me why its not changing the password. is it a bug in this apllication or is there something which i have to do to make it work? thanx in advance... -- Regards Rizwan Hisham Software Engineer ___ --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] Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls - Transfer # Not Wor king
In the extensions.conf do you have:
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,tT) ?
for the outgoing calls?
regards,
Staalenburg, Juan escribió:
Trixbox 1.2.3 - TDM400 FXOs - Flash (*) and # Not Working
Has anyone run into this problem. I cannot transfer or park a call (#) on
outgoing calls. Using Zaptel TDM400 FXO card. This may be normal but I
wanted to check.
Regards,
Juan S.
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--
José Luis Ledesma
Tecnobe Tecnología S.L.
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[asterisk-users] Friday asterisk users live conference/podcast at 12:30PM EDT
I am starting an asterisk users live conference call on Talkshoe, a robust voIP conferencing platform I use for several podcasts. Although I have spoken to Mark Spencer and a Digium VP about this idea, they have nothing to do with it for the moment. They may wish to come on board later if enough people show interest. Further disclaimer, there is NO commercial intent behind this initiative. I only hope to bring members of the user community together. Should you care to take part as a speaker, you need only open a free Talkshoe account. That account will provide you with the needed 10 digit PIN to speak at the conference. You can call in via SIP or (US) PSTN. There is a text chat client you can optionally download to both listen and add text comments or questions without calling in. You can listen to either live or recorded episodes without subscribing or joining anything at all. Just visit the show page http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 More information links: http://pages.x2z.eu Your are most welcome to add any comments here or email me directly with any questions. I hope to hear you there beginning Friday, March 30, 2007 at 12:30PM EDT! Regards, wp ___ --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] Doorphone vs. Grandstream BT101
Jay, Just for the record, I own 3 BT102 and all three have stopped working for various different reasons. This make me think that um... they're not very good. Two had hardware problems, one of those was minor (handset cord) and one will not work no matter what firmware I use. Grandstream tried to help debug it but it wouldn't stay registered so all three are now doorstops for three different problems. ITH, I have three different IAX phones that all work perfectly for $80 or less. ___ --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: SIP Video Camera
KokMengLoh wrote: Hi, Does anyone know of a Video Camera that is based on SIP? There are lots of Video Phones out there, but I can't seem to find a Video Camera. What would you do with SIP video camera? -- Tomislav Parcina [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
[asterisk-users] System from AMI
How to execute some system command from AMI? -- Tomislav Parcina [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
[asterisk-users] Re: AOC billing
Stefano Corsi wrote: is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Ciao Stefano! Since I'm not a programmer I'm waiting for some AOC solution, but nothing was developed so far. -- Tomislav Parcina [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] Re: AOC billing
At 14.02 28/03/2007, you wrote: Stefano Corsi wrote: is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Ciao Stefano! Since I'm not a programmer I'm waiting for some AOC solution, but nothing was developed so far. So there's no way to use it in the CDR logs? Is there some other way, apart from maintaining call rate tables, to get direct billing information from the telco? Thanks Rgds 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
[asterisk-users] wireless desktop phones
I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? ___ --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: just on my LAN
Josu Lazkano Lete wrote: hello I want to install Asterisk just to use in my LAN, without a analog or digital devices. I need to install all this packages??? Asterisk 1.2.17 http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz Zaptel 1.2.16 http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz Libpri 1.2.4 http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz Addons 1.2.5 http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.5..tar.gz Sounds 1.2.1 http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1..tar.gz You realy should read this http://www.voip-info.org/wiki-Asterisk -- Tomislav Parcina [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] wireless desktop phones
Yeh Jordan, my suggestion is don't. If you read this list you'll find plenty of people complaining about wireless functionality, the hardware/technology just isn't there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Wednesday, 28 March 2007 8:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] wireless desktop phones I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] * 1.4.1: connected to gtalk but no voice passing
Hi, I managed to connect Asterisk 1.4.1 to my gtalk account but after calling I hear no voice from other side (a SIP phone). Asterisk log says nothing. What am I missing? TIA Giorgio Incantalupo ___ --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: Re: Inbound Voice Quality - Speed Change
Yes, this is the output from the lsmod. I should have posted that for clarification. I was assuming that asterisk would have used the ztdummy module and the lsmod command would have indicated that at least 1 program had opened the driver interface. I'm reading more about ztdummy now to see if anything else is required, for example, udev configuration. I use Fedora Core 5. Jim Travis Schafer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Looks like output from the 'lsmod' command. Lacy Moore - Aspendora [EMAIL PROTECTED] 3/27/2007 11:34 PM On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: ztdummy 4424 0 rtc11156 1 ztdummy zaptel178084 1 ztdummy crc_ccitt 2016 1 zaptel Ok, this is a dumb question, but what is that output from? What distribution of Linux are you using? I've never had to change anything related to the kernel. I use CentOS, though. ___ --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] wireless desktop phones
Any comments on an ATA and an analog wireless? I've been doing it that way and it works well... Todd On Mar 28, 2007, at 8:31 AM, Dean Collins wrote: Yeh Jordan, my suggestion is don’t. If you read this list you’ll find plenty of people complaining about wireless functionality, the hardware/technology just isn’t there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. ___ --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] Cisco 30VIP Phone
Chris Nighswonger wrote: Is anyone else on the list using Cisco 30VIP phones with the chan_skinny driver? I have tried to catch the one of the developers on the chat relay, but cannot seem to get anywhere. I am trying to understand how the soft buttons are setup. They are apparently hard-coded into the chan_skinny.c module. Specifically, I am looking for how the code relates to the actual layout of the buttons on the phone. So far, I cannot even get the buttons that are in the code by default to work properly. I have several of these phones up and registered with *. The dialpads work fine. But other buttons do not. Thanks Chris On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have three registering with * and having basic functionality. I am at a loss to know how to program the buttons (other than dtmf, hold, mute, spkr). Here is what the * console shows when one of the phones registers: -- Starting Skinny session from 192.168.0.70 -- Device 'SEP000196C00CDC' successfully registered Device capability set to '12' Adding button: 9, 1 Adding button: 1, 0 Adding button: 15, 0 Adding button: 126, 0 Adding button: 5, 0 Adding button: 125, 0 It appears that * is setting up some buttons. But where it is getting the config info, I don't know. Sorry for answering my own post, however it may help someone else: Soft button configuration is set in skinny.c I'm still looking for some explaination of the logic and sytax of setting them. Chris if i remember right, most of the buttons on those and the 12SP+ phones don't really work because there isn't a button template in * signature.asc Description: OpenPGP digital 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] wireless desktop phones
On Wed, 28 Mar 2007, Dean Collins wrote: Yeh Jordan, my suggestion is don't. If you read this list you'll find plenty of people complaining about wireless functionality, the hardware/technology just isn't there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. I would add to this by saying the same... (Assuming you're talking about Wi-Fi) The technology is there, but I'm not convinced it's robust enough - yet. I'm sure it will get there though. Wi-Fi has many issues - including performance - with many subscribers to a single base-station you'll experience drop-outs, packet loss, etc. However, if you're looking for wireless, then you might want to look at some of the DECT solutions - either by connecting analuge base stations to a TDM card, or using a SIP compatable base station. I've just deployed a pair of Siemens CP460IP's and just ordered a couple more. So-far so good. They aren't perfect - check the WiKi for some details though. http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP And if you need to extend range, there are relay units avalable, although I've found coverage to be better than other DECT systems I've used. The down-side is that you can only (I think) have 6 base stations in any one area, so if you're looking to give everyone their own wireless phone, it may prove to be problematic - however I've not got the hard facts on number of DECT basestations, so I could be wrong here. On the WiFi side, the only phone I've played iwth is the UT Starcom F1000G, and while it works, most of the time, it's a bit too geeky for general use - it didn't pass the wife test... Good luck Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Wednesday, 28 March 2007 8:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] wireless desktop phones I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? ___ --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] wireless desktop phones
Aastra has some new products coming that combine DECT with SIP, and look promising. Linksys also makes an 802.11G WIFI dongle that can be mated with their SPA-9XX series phones to untether them from your wired LAN, and have no direct feedback on these in a commercial deployment however. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd H Sent: Wednesday, March 28, 2007 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wireless desktop phones Any comments on an ATA and an analog wireless? I've been doing it that way and it works well... Todd On Mar 28, 2007, at 8:31 AM, Dean Collins wrote: Yeh Jordan, my suggestion is don't. If you read this list you'll find plenty of people complaining about wireless functionality, the hardware/technology just isn't there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. ___ --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] Can I generate random SIP traffic?
Hi Gabriele, maybe sipp can help you: http://sipp.sourceforge.net/ Giorgio [EMAIL PROTECTED] wrote: Hello, I would like to generate a peer-to-peer or a server/client SIP traffic between two or more Openwrt access point, to make some statistics about QoS. I tried some SIP traffic generators for OpenWrt, but I didn't find nothing of satisfactory. Now I wonder if asterisk can help me generating random SIP traffic. I'm googling since yesterday without results. Can you help me plz? Thanks and sorry for the disturb. ___ --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] Doorphone
Ola Lidholm wrote: In queue.conf (or is it called queues.conf?) you can set up a call queue with all your phones already in it. Which will mean that if you pass the incoming call to that queue all phones will be ringing until one person picks it up. At my work we have it set up like that. And additionally, people can join or leave the call queue by dialing certain extensions on their phones, which can be convenient when people do not want to be disturbed. I do not understand exactly how you mean your system works, how does the users know when someone is at the door? Since no phone is ringing it seems to me like a guessing game to know when they need to dial in to the meetme to open the door? Do you have free sight to the entrance door so that you can see if someone is already there? /Ola LOL. There's a door chime that rings that everyone can hear, and there's a second or two after it goes off where we all look at each other to see who's gonna budge. ___ --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] wireless desktop phones
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all wireless so the phones would have all the bandwidth.
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[asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm
Meetme cant handle more than 5 users in a call?? H http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/ hmmm I'm all for commercializing a product, but this FUD from Fonality seems to be taking it just a little too far Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph ___ --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] MOS Score
Does anyone know of free/cheap/open source software that will allow me to run a test for a period of time and get an MOS score for 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
[asterisk-users] h323
hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI -- Executing Dial(SIP/2.2.2.2-086f5ac0, H323/[EMAIL PROTECTED]|60) in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28 14:17:23 NOTICE[11985]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/2.2.2.2-086f5ac0' status is 'CHANUNAVAIL' Don't get soaked. Take a quick peek at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather ___ --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] wireless desktop phones
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Wednesday, 28 March 2007 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wireless desktop phones On Wed, 28 Mar 2007, Dean Collins wrote: Yeh Jordan, my suggestion is don't. If you read this list you'll find plenty of people complaining about wireless functionality, the hardware/technology just isn't there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. I would add to this by saying the same... (Assuming you're talking about Wi-Fi) The technology is there, but I'm not convinced it's robust enough - yet. I'm sure it will get there though. Wi-Fi has many issues - including performance - with many subscribers to a single base-station you'll experience drop-outs, packet loss, etc. However, if you're looking for wireless, then you might want to look at some of the DECT solutions - either by connecting analuge base stations to a TDM card, or using a SIP compatable base station. I've just deployed a pair of Siemens CP460IP's and just ordered a couple more. So-far so good. They aren't perfect - check the WiKi for some details though. http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP And if you need to extend range, there are relay units avalable, although I've found coverage to be better than other DECT systems I've used. The down-side is that you can only (I think) have 6 base stations in any one area, so if you're looking to give everyone their own wireless phone, it may prove to be problematic - however I've not got the hard facts on number of DECT basestations, so I could be wrong here. On the WiFi side, the only phone I've played iwth is the UT Starcom F1000G, and while it works, most of the time, it's a bit too geeky for general use - it didn't pass the wife test... Good luck Gordon, If you need to have high density DECT then it's very easily achievable but like all things you need to move to a commercial situation rather than a domestic style gigaset. I used to sell commercial DECT solutions (eg starting at $40k+ - my largest individual site was a 250K solution). I've even seen an ericsson md110 with nothing but high density dect cards supporting a multi acre military facility. My point is you can have more than 6 handsets in a single 'zone'. To answer Jordan's original question - why do you want wifi? Do these people have desks? Monitors? Pc's connected to cables? Then don't be silly and try to give them wifi when the technology is too immature. If you ARE working in a trading situation where people don't have desks and are totally mobile then you need to use commercial DECT. 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
Re: [asterisk-users] wireless desktop phones
Jordan Novak wrote: I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable I've been looking at 802.11g wireless 8 port switches. I have run into a few hits on Google, that may help. http://www.dlink.com/products/?pid=434sec=0 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
Re: [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm
Yikes! While I will agree I think Digium needs to do a little better QA (let's not start that war again), this kind of FUD doesn't do anything for the community. I've had Asterisk running with meetme no problem with many more then 5 users. On 3/28/07, Dean Collins [EMAIL PROTECTED] wrote: Meetme cant handle more than 5 users in a call?? H http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/ hmmm I'm all for commercializing a product, but this FUD from Fonality seems to be taking it just a little too far Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph ___ --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] Doorphone vs. Grandstream BT101
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Tuesday, March 27, 2007 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone vs. Grandstream BT101 This is the simplest solution I can think of: http://www.smarthome.com/5070cw.html On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote: Steve Totaro wrote: Just get a Grandstream ATA and a handset with no buttons. So simple. That doesn't really meet my needs -- I want to be able to dial-out, and have the person on the other end simply be able to push a button to ring the doorbell. The doorbell button requirement stems from the eternal hope that someday DHL drivers will be trained to push it just before or after they slam-dunk that box marked fragile, so I can get this box of broken computer parts out of the pouring rain when it arrives, and won't have to file the insurance claim days later when I find the rain-soaked cardboard blob near the culvert. Sorry, I got sidetracked there. What I mean is, I'd like to keep the doorbell button so that Fedex and UPS drivers can continue to ring it and leave when they deliver something -- I think having to pick up a crusty, dirty receiver might be a deterrent even to those 99.9% of folks who are better trained than a DHL driver. Shoot, went there again. No I'm not bitter. OK Anyways... You could still use a Grandstream ATA and just have your doorbell switch actually be the hook switch for the line, use the h extension to continue ringing phones, send an SMS, jabber message or whatever. Just set the auto dial in the ATA. Or you could just use a regular old doorbell or one of the wireless units sold at Radio Shack, Sears, Wal-Mart, and everywhere. It pains me to say it, but not everything needs to integrate with Asterisk. Sometimes a doorbell should just be a doorbell, why make things more complicated than they need to be? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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] ztdummy and MOH
Klaverstyn, David C wrote: I am using autoload and I have rebooted the server. I have tried using different files and a different location. This is getting very frustrating. If the call was a SIP call then I would say that the device is using VAD/CND (silence detection). This is the classic cause of MoH only working when there is audio going in the other direction. Maybe there is SIP somewhere in the call path. -- This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1175091047): Split unusually long word(s) in header. SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): Enforced policy: accept Total modifications so far: 1 Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.94 2006/01/02 16:43:10 bre Exp $ ___ --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] System from AMI
Tomislav Parcina wrote: How to execute some system command from AMI? You have to login into the AMI server with proper credentials and send commands. I wrote an AMI test application a little while back. It gives you the ability to login into the AMI, send commands and snoop packets being send out. Great way to get familiar with AMI commands and packet structure. http://www.datatrakpos.com/pos/datatalk/dpdownload.aspx You'll see the download under Manager API Testing Utility. It's freeware. -- Warm Regards, Lee ___ --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: Re: Inbound Voice Quality - Speed Change
This is just a guess. I suspect the use count is counting the number of kernel modules that are using another kernel module. Sort of a depends on thing. i.e. zttdummy is using rtc and zaptel. zaptel is using crc_ccitt. Since Asterisk is not a kernel module and it access Zaptel via syscalls or opening up files in /dev or /proc, it would not be counted in the use count. Jim Duda wrote: Yes, this is the output from the lsmod. I should have posted that for clarification. I was assuming that asterisk would have used the ztdummy module and the lsmod command would have indicated that at least 1 program had opened the driver interface. I'm reading more about ztdummy now to see if anything else is required, for example, udev configuration. I use Fedora Core 5. Jim Travis Schafer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Looks like output from the 'lsmod' command. Lacy Moore - Aspendora [EMAIL PROTECTED] 3/27/2007 11:34 PM On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: ztdummy 4424 0 rtc11156 1 ztdummy zaptel178084 1 ztdummy crc_ccitt 2016 1 zaptel ___ --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] Development of new features in Asterisk Manager
Hi, Has anybody customized* anything in Asterisk? * Customized = Development of new features or changes the existent features. I need a new feature in Asterisk Manager and would like to talk about this. Thanks, Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil ___ --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] Couldn't load variables.txt?aldope=xxxxx
Hi Giorgio, sorry but how do this? how i verify the server it's running, and if not runnig how i put this running. Thanks 2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]: Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? ___ --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 -- Carlos Jerónimo ___ --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] wireless desktop phones
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire
30% of these end-points. P{hysically impossible. They do have cat3
twisted pair to each phone. But of course they want IP. Are there any
adpaters that will give me just enough bandwidth to get it done. The
computer network is all wireless so the phones would have all the
bandwidth.
You should be able to use 10baseT over Cat 3 cables.
A half decent manageable 10/100 switch will be able to lock the ports to
10Mb. 10 megabits is plenty for voice.
Depending on the quality of the wiring, your mileage may vary. Take care
to use proper structured wiring techniques. Put Cat5 sockets on the desk
end and a patch panel in the closet. DO NOT crimp RJ45 connectors onto
building wiring. They are not meant for this type of cable and THEY WILL
NOT BE RELIABLE.
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com
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Re: [asterisk-users] Inbound Voice Quality - Speed Change
Could it possibly be a packetization rate issue with your provider? On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound Voice Quality - Speed Change
And/or periods of large jitter on your network connection. On 3/28/07, Matt [EMAIL PROTECTED] wrote: Could it possibly be a packetization rate issue with your provider? On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PoE - IEEE 802.3af
Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Doorphone
Responsibility for answering the door is shared by the entire office. But A) noone wants their phone to ring, there's a door chime) and B) noone specific will accept responsibility for answering the door. So, we need a solution that follow I'm answering the door now, these are the buttons I push. So, when someone is at the door, you call whatever extension to get to the door intercom, talk to them, then you decide to open it. You hangup, then dial an extension that does only this, unlock the door. Something like [door-opener] exten = 555,1,System(script_to_unlock_door.sh) exten = 555,n,Hangup() If you really don't want to have to dial a second extension, look at applicationmap in features.conf http://www.voip-info.org/wiki/view/Asterisk+config+features.conf hth ___ --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] Development of new features in Asterisk Manager
On Wed, 2007-03-28 at 11:20 -0300, Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote: Hi, Has anybody customized* anything in Asterisk? * Customized = Development of new features or changes the existent features. I need a new feature in Asterisk Manager and would like to talk about this. This is a good place to discuss it; what you want to do, may already have been done in dozens of places! And people love to brag! :) murf Thanks, Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil -- Steve Murphy Software Developer Digium 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] Using server side phonebook directory with SPA941
On 3/27/07, Robert Lister [EMAIL PROTECTED] wrote:
On Tue, Mar 27, 2007 at 12:45:44PM +0200, Maxim Veksler wrote:
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that when this party calls a textual caller ID will
be displayed on the phone display.
Can be done reasonably easily in the dial plan. What I have is quite noddy
but it does the job. In the incoming bits of dial plan where calls come in,
I call this as a macro in the context where incoming calls arrive, before
handing it off to the Dial() bits:
exten = _4535XX,1,Macro(setisdncallerid,${EXTEN},PSTN,9)
What this macro (pasted below) does is allow alpha tagging of incoming
calls, plus some defaulty stuff set by the gateway (caller ID not
present/withheld comes through in my case as either anonymous or just 0 or
00, so this macro tidies this up before passing the call on.)
It also inserts the access digit (9) in front of the caller ID as in my case
outside calls need a 9 prefix. This is just so that call routing works
correctly if people return missed calls/save numbers from the handset etc.
Obviously you will have to tweak this for your setup.
If there is no alpha tag in the DB, it sets some defaulty thing (In my case
PSTN to give some indication where the call is coming from.)
It can also do a CPI tag based on destination number, for queues/group
numbers, so that the alpha tag on the call gets set to something like
Main Number etc. to distinguish a DDI call from a Queue Call.
The database entries look like:
*CLIdatabase put tag 01234567890 Some Name Here
and for CPI (called party) Tag:
*CLIdatabase put 453510 tag Helpdesk
[macro-setisdncallerid]
; ${ARG1} = Called Party Number (XX) as presented from BT.
; ${ARG2} = default tag to add to incoming calls
; ${ARG3} = prefix to insert to incoming CLI
;
; Frobs the incoming caller ID headers how we like it:
exten = s,1,NoOp(macro-setisdncallerid: ${ARG1})
; In my case the internal extension is 7XX where XX is the
; last two digits of the incoming DDI number. This just makes
; it display right in the caller ID:
exten = s,2,Set(DIALED_EXTEN=7${ARG1:-2})
; For cisco phone, set different ring cadence to indicate
; an external call:
exten = s,3,SIPAddHeader(Alert-Info: Bellcore-dr2)
exten = s,4,GotoIf($[ ${CALLERID(num)} = anonymous ]?400)
exten = s,5,GotoIf($[ ${CALLERID(num)} = 0 ]?500)
exten = s,6,GotoIf($[ ${CALLERID(num)} = 00 ]?500)
exten = s,7,GotoIf($[ ${DB(tag/${CALLERID(num)})} != ]?700)
exten = s,8,Set(CALLERID(name)=${ARG2} to ${DIALED_EXTEN})
exten = s,9,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,10,Goto(900)
exten = s,400,Set(CALLERID(name)=${ARG2})
exten = s,401,Goto(900)
exten = s,500,Set(CALLERID(num)=unknown)
exten = s,501,Set(CALLERID(name)=${ARG2})
exten = s,502,Goto(900)
exten = s,700,Set(CALLERID(name)=${DB(tag/${CALLERID(num)})})
exten = s,701,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,702,Goto(900)
; If there is a CPI tag set, use that: (i.e. SUPPORT)
exten = s,900,GotoIf($[ ${DB(${ARG1}/cpitag)} != ]?950)
exten = s,950,Set(CALLERID(name)=${DB(${ARG1}/cpitag)})
2. How can this be configured with Trixbox, I've looked at the
configuration options - I assume it plays no difference me basing it
on mysql or astdb?
3. What protocol does the phone (Linksys SPA941) talks to the
asterisk server to retrieve this information ?
When an incoming call arrives with asterisk, the SIP headers can be set
appropriately before you present this information to the handset. It's in
the incoming SIP packets to the handset.
4. Has someone done this? What softphone should I use to test it first
(I'm connecting it with outlook, so it has to be win* software)
There are a few to choose from. I use Counterpath's X-Lite client:
http://www.counterpath.com/
Thank you Rob for the detailed reply.
It solves one side of the problem (In a very cool and unexpected way I
must admit) but not the whole demand. I still would like to have a
centrally managed caller phonebook directory, available from the
phone's Directory menu. I did found some solutions[1] involving a
push method with wget to each phone. It slick, but it requires some
API between a php script and asterisk to query for registered devices,
then making a push to them.
see,
[1] http://grimsy.blogspot.com/2007/02/spa942-personal-directory-ldap.html
Rob
Maxim.
--
Cheers,
Maxim Veksler
Free as in Freedom - Do u GNU ?
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Re: [asterisk-users] PoE - IEEE 802.3af
A POE switch will put power on what ever line is connected to it, so if your polycom plugs into a wall plate with cat 5 cable that runs back to a port on the POE switch then you have power all the way to the phone. On 3/28/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3afcable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves 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] wireless desktop phones
Aastra just released a DECT SIP solution. Supposedly they are the first to do so but who knows. I'm not affiliated with them so it's not a plug or anything. http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-12C98649/04/hs.xsl/21410.htm -Original Message- From: Gordon Henderson [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 28, 2007 5:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wireless desktop phones On Wed, 28 Mar 2007, Dean Collins wrote: Yeh Jordan, my suggestion is don't. If you read this list you'll find plenty of people complaining about wireless functionality, the hardware/technology just isn't there yet. Stick with wired phones and one or two wireless for particular people for now, maybe in 12-18 month things might change. I would add to this by saying the same... (Assuming you're talking about Wi-Fi) The technology is there, but I'm not convinced it's robust enough - yet. I'm sure it will get there though. Wi-Fi has many issues - including performance - with many subscribers to a single base-station you'll experience drop-outs, packet loss, etc. However, if you're looking for wireless, then you might want to look at some of the DECT solutions - either by connecting analuge base stations to a TDM card, or using a SIP compatable base station. I've just deployed a pair of Siemens CP460IP's and just ordered a couple more. So-far so good. They aren't perfect - check the WiKi for some details though. http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP And if you need to extend range, there are relay units avalable, although I've found coverage to be better than other DECT systems I've used. The down-side is that you can only (I think) have 6 base stations in any one area, so if you're looking to give everyone their own wireless phone, it may prove to be problematic - however I've not got the hard facts on number of DECT basestations, so I could be wrong here. On the WiFi side, the only phone I've played iwth is the UT Starcom F1000G, and while it works, most of the time, it's a bit too geeky for general use - it didn't pass the wife test... Good luck Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Wednesday, 28 March 2007 8:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] wireless desktop phones I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? ___ --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] PoE - IEEE 802.3af
Mike wrote: Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike Yes it is true that you need a special cable (provided in the 802.3af bundle). This cable is black, about 4 ft long and has an RJ45 plug (goes into the phone) on one end and an RJ45 Jack (plugs into the patch cable) on the other. Near the RJ45 jack there is a little black box that does the power conversion from 802.3af into whatever the phone needs. There is also a (not sure what it's called) spot for you to plug in an AC adapter if necessary. There's a good picture of one here: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-38040890624.htm As long as you have CAT5 or better cable between the switch and the RJ45 jack on the special cable you will be fine. Using a patch panel and a wall jack is also fine as long as it is CAT5 or better. You just can't have another switch or hub between your POE switch and the phone. -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
[asterisk-users] Asterisk: recommended installation
Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does [EMAIL PROTECTED] or Trixbox match to my scenario By the way, I use Debian Etch as OS server. Really thanks. Alejandro -- Alejandro Cabrera Obed Interconexion SINTyS Sistema de Identificación Nacional Tributario y Social Consejo Nacional de Coordinación de Políticas Sociales Presidencia de la Nación Julio A. Roca 782 - Piso 5 Ciudad Autónoma de Bs. As. Tel: (54 11) 4343-0181/89 interno 5172 4334-3676 4342-5648 [EMAIL PROTECTED] NOTA DE RESPONSABILIDAD: -- Este mensaje proviene de Internet,tome los recaudos necesarios en su manejo. El contenido del presente mensaje y sus adjuntos es privado, estrictamente confidencial y exclusivo para su destinatario, pudiendo contener información protegida por normas legales y de secreto profesional. Bajo ninguna circunstancia su contenido puede ser transmitido o revelado a terceros ni divulgado en forma alguna. En consecuencia de haberlo recibido solicitamos contactar al remitente y eliminarlo de su sistema. -- ___ --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] PoE - IEEE 802.3af
You don't need to change any wiring. Just be sure that the LAN wiring terminates at a PoE LAN switch (PoE would not be passed through an intermediate switch). You will get an AC adapter with your phone. If the phone fails to power up, you can plug the adapter into the thingie in the PoE cable (not the phone). Also, the IP601 has a 24V AC adapter while the IP501 has a 12V adapter. Mike wrote: Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided special RJ-45 cable, into a PoE capable switch, and voilà! Is this true? And if so, what happens when the Phone doesn't connect directly to the switch? (let`s say there is wiring in the wall that goes to a patch panel, for example. Do I need to change all the wiring in the office?) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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] wireless desktop phones
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all wireless so the phones would have all the bandwidth.
Some of the Wifi phones--at least under the relatively stable conditions
I have here--work very reliably.
I have 3 Starcom F1000s, and a) if they don't have to roam and b) they
don't have to connect dynamically to different servers, work just fine.
FYI. YMMV.
B.
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Re: [asterisk-users] TDM400p reliability
Hi Joe - What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? Back to the original topic... I have 6 of these cards installed in various asterisk installations (used by businesses), the oldest of which is about 2.5 years. There are a varying mixture of FXO and FXS ports on these cards. I haven't seen any failures. All of them have just worked. I saw a complaint in this thread: they can be something of a nightmare to get right with echo issues and the like. Keep in mind that if the quality of the signal on your phone lines is good, you are unlikely to have echo problems. Of the 6 cards I mentioned above, I put 5 of them in place and never had an echo issue. On the 6th, it was an older building with really crappy wiring, and I did see some echo issues. I recently installed HPEC there, and with no tuning at all, it has totally solved all problems. - Noah ___ --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] Dialplan Streaming
All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. Doug. ___ --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] BRI Cards
Hi all, I am looking for a reliable BRI (8 port) card, and I wonder which BRI card would you guys recommend me to use? The card will have to work in a PCI slot that is sharing IRQ with another device...does that represent a problem (and if so, for which cards)? Regards, Alex ___ --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] PoE - IEEE 802.3af
Hi Mike - You don't need to change any wiring. Just be sure that the LAN wiring terminates at a PoE LAN switch (PoE would not be passed through an intermediate switch). One little caveat: Depending on the PoE mode, you may need to use all four pairs of the Cat 5 cable of your network wiring. I'm not sure which mode the Polycom PoE cables prefer, but Mode A PoE uses the same two pairs (1-2, 3-6) as 100BASE-TX Ethernet, while Mode B PoE uses the other two pairs (4-5, 7-8). Most good network wiring will use all four pairs, but some wiring installations will only use two pairs, so it's probably a good idea to just double-check that yours uses all four pairs. - Noah ___ --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: Friday asterisk users live conference/podcast at 12:30PM EDT
Further disclaimer, there is NO commercial intent behind this initiative. I only hope to bring members of the user community together. Someone kindly emailed privately about this. By the above disclaimer I mean that I myself have nothing to sell in doing this not is it meant to be a marketplace. However, anyone who has solutions or services is welcome to talk about them as time permits assuming they are asterisk related and pertinent and interesting to the user community. The less rules the better. The conference is moderated and will not degenerate into flames or sales pitches. wp ___ --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] MOS Score
Matt wrote: Does anyone know of free/cheap/open source software that will allow me to run a test for a period of time and get an MOS score for VoIP? This one is great: http://www.testyourvoip.com Its free and you can use it all you want. If you want to buy it to install on your server, its quite expensive though. ___ --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 -- Andres Technical Support http://www.telesip.net ___ --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: Question about DSP in Digium card
On Mar 27, 2007, at 8:35 AM, Salvatore Giudice wrote: As for the DSP, you are right to be concerned about the Digium cards, but not because of the DSP. The DSP is not where you will run into problems. Digium cards feature 2 year old circuitry and do not play well with other devices. You have to take care not to share interrupts with any components that may be active on that system. Sharing an IRQ between a Digum card and an Ethernet card would certainly spell disaster in my experience. From personal experience, I no longer use Digium hardware since I could rarely push a quad port card to more than 13 channels per T1 circuit without the card failing miserably. HDLC aborts abound. FWIW, there have been some recent improvements in the drivers and firmware which correct most of the old IRQ sharing and HDLC problems of that nature. If you have any more such problems, be sure to let tech support know so we can get it fixed. We are anxious to keep your business. Matthew Fredrickson ___ --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] Using server side phonebook directory with SPA941
Maxim Veksler wrote: Thank you Rob for the detailed reply. It solves one side of the problem (In a very cool and unexpected way I must admit) but not the whole demand. I still would like to have a centrally managed caller phonebook directory, available from the phone's Directory menu. I did found some solutions[1] involving a push method with wget to each phone. It slick, but it requires some API between a php script and asterisk to query for registered devices, then making a push to them. see, [1] http://grimsy.blogspot.com/2007/02/spa942-personal-directory-ldap.html When using SIP it is up to the phone to handle directories, not the SIP server. Polycom has support for centrally managed directories, but it has nothing to do with the SIP part of the phone. It is considered a provisioning 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
Re: [asterisk-users] Question about DSP in Digium card
Hi Steve - Just my personal experience, but I do not find IAX to be very reliable. Is there any particular reason you are not using SIP? I'm curious as to your negative experiences with IAX. I generally use it for multi-office installations, and have had good expereinces with it. What reliability issues did you see? Jitter? Drops? Thanks, Noah ___ --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] Doorphone vs. Grandstream BT101
Steve Totaro wrote: OK Anyways... You could still use a Grandstream ATA and just have your doorbell switch actually be the hook switch for the line, use the h extension to continue ringing phones, send an SMS, jabber message or whatever. Just set the auto dial in the ATA. I got a grandstream on order, so I'll try that out. Or you could just use a regular old doorbell or one of the wireless units sold at Radio Shack, Sears, Wal-Mart, and everywhere. It pains me to say it, but not everything needs to integrate with Asterisk. Sometimes a doorbell should just be a doorbell, why make things more complicated than they need to be? I currently do have a good ole doorbell, and a $10-direct-from-hong-kong wireless doorbell, with the transmitter triggered by the existing solenoid via reed-switch. However, we're finishing the walk-up attic this year, moving my office up on the third floor, so an intercom is in order. In that context it makes perfect sense to use the existing phone system and integrate the doorbell, wouldn't you agree? ___ --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] ztdummy and MOH
On 3/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Klaverstyn, David C wrote: I am using autoload and I have rebooted the server. I have tried using different files and a different location. This is getting very frustrating. If the call was a SIP call then I would say that the device is using VAD/CND (silence detection). This is the classic cause of MoH only working when there is audio going in the other direction. Maybe there is SIP somewhere in the call path. I have the similar problem on 1.4.1. I don't remember having it in 1.4.0, I could be wrong. I have a SIP provider, when calls come in, it play MOH while waiting for to be picked up. ztdummy is loaded. Another interesting thing I notice, exten = s,1,Zapateller(answer|nocallerid) exten = s,n,Background(PleaseWait) exten = s,n,Dial(100,30,r) Please note, if I use r (ring) instead of m in the Dial option, I have choppy ring too. If I rub my finger on the mouth piece, the ring/MOH is fine. Any solution to this problem? I'm using asterisk 1.4.1 with zaptel 1.4.0. Thanks. Wooi ___ --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: How is this feature called ?
No, I'm far from inventing features, yet ! ;-) It's a feature offered by Alcatel and I wanted to find in documentation, a way to reproduce it, just in case I'm asked to do so. I think it's the equivalent of call screening, but from caller perspective. 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
RES: [asterisk-users] Development of new features in Asterisk Manager
Hi Murphy, I am developing an application for integration with Asterisk by Asterisk Manager. When I send a command to asterisk (Example: Action: Originate), many events are raised. I would like to identify what events answer my command. I'm thinking of creating a new property in the events to return the command ActionID. Example: I send an Originate: action: Originate actionid: 1234526_PS channel: local/092053469 Exten: 101 Context: default Priority: 1 Async: true Many events are returned: (In this case didnt answer the call) Event: Newchannel Privilege: call,all Channel: Local/[EMAIL PROTECTED],1 State: Down CallerIDNum: unknown CallerIDName: unknown Uniqueid: 1175078296.23 Event: Newchannel Privilege: call,all Channel: Local/[EMAIL PROTECTED],2 State: Ring CallerIDNum: unknown CallerIDName: unknown Uniqueid: 1175078296.24 . . . Event: Dial Privilege: call,all Source: Local/[EMAIL PROTECTED],2 Destination: SIP/tmaisMG-096eee20 CallerID: unknown CallerIDName: unknown SrcUniqueID: 1175078296.24 DestUniqueID: 1175078296.25 . . . Event: Hangup Privilege: call,all Channel: Local/[EMAIL PROTECTED],2 Uniqueid: 1175078296.24 Cause: 16 Cause-txt: Normal Clearing Event: Hangup Privilege: call,all Channel: Local/[EMAIL PROTECTED],1 Uniqueid: 1175078296.23 Cause: 0 Cause-txt: Unknown Event: OriginateResponse Privilege: call,all ActionID: 1234526_PS Response: Failure Channel: local/092053469 Context: default Exten: 101 Reason: 1 Uniqueid: null CallerID: unknown CallerIDNum: unknown CallerIDName: unknown Only OriginateResponse has ActionID. How can I identify the Newexten, Newchannel, Hangup, Dial, , source? Do you understand me? Thanks, []s Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Steve Murphy Enviada em: quarta-feira, 28 de março de 2007 12:36 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Development of new features in Asterisk Manager On Wed, 2007-03-28 at 11:20 -0300, Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote: Hi, Has anybody customized* anything in Asterisk? * Customized = Development of new features or changes the existent features. I need a new feature in Asterisk Manager and would like to talk about this. This is a good place to discuss it; what you want to do, may already have been done in dozens of places! And people love to brag! :) murf Thanks, Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil -- Steve Murphy Software Developer Digium ___ --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] PoE - IEEE 802.3af
Thanks for all the replies, this definitely helps me! Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, March 28, 2007 12:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE - IEEE 802.3af Hi Mike - You don't need to change any wiring. Just be sure that the LAN wiring terminates at a PoE LAN switch (PoE would not be passed through an intermediate switch). One little caveat: Depending on the PoE mode, you may need to use all four pairs of the Cat 5 cable of your network wiring. I'm not sure which mode the Polycom PoE cables prefer, but Mode A PoE uses the same two pairs (1-2, 3-6) as 100BASE-TX Ethernet, while Mode B PoE uses the other two pairs (4-5, 7-8). Most good network wiring will use all four pairs, but some wiring installations will only use two pairs, so it's probably a good idea to just double-check that yours uses all four pairs. - Noah ___ --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] Transfering not working - how to debug?
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
these buttons on my handset nothing happens (other than I hear the dtmf
tones on the other end of the line).
roo*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Attended Transfer #2
One Touch Monitor *1
Disconnect Call * *0
I am using the tT options in my dial calls (via a macro)
[macro-extension]
exten = s,1,Dial(${ARG1},20,tT)
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] wireless desktop phones
The RFP 32 access point that comes with Aastra solution reminds a product sold by DeTeWe, a company Aastra bought months ago. At that time, I thought it was a Kirk OEM but I've got no elements proving it (just by looking at both products). 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
Re: [asterisk-users] SRTP vs ZRTP in Asterisk
Do you mean it c(sh)ould be included in 1.6 ? ;-) 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
[asterisk-users] Multi-line phones - Asterisk uses wrong callerid
I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like to resolve:- 1. The person called sees the wrong callerid 2. The CDR records the call against the wrong account 3. Picking up voicemail requires multiple extra steps Is there a way around this?? Scenario:- Phone 1 has three lines 101, 102, 103 Phone 2 has 1 line 202 User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) User 2 at Phone 2 sees call coming from extension 103 instead of 101 With 'sip debug' enabled at the console, I see an INVITE issued (on the Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xx mailbox=101 callerid=User 1 101 sip show peers :- 103/10310.10.10.181 D 5060 OK (157 ms) 102/10210.10.10.181 D 5060 OK (159 ms) 202/20210.10.10.184 D 5060 OK (4 ms) 101/10110.10.10.181 D 5060 OK (160 ms) Asterisk 1.2.15 Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfering not working - how to debug?
Alan Chandler wrote: I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit You need to also include the t and/or T in your dial statement. 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
Re: [asterisk-users] wireless desktop phones
Jordan Novak wrote: I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? If you enjoy being miserable and having your phones not work, by all means, use a wi-fi phone. Frankly, it's worth the extra coin to get a cable run done to wherever you need it. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Inbound Voice Quality - Speed Change
Matt, That's possible. I've been struggling with this for a while. I recently transitioned from cable modem service to Verizon FIOS. I didn't get a big change in behavior ( I was hoping so ). My VOIP provider is Teliax. My ping responses to the Teliax server are around 13/15 mS. Can you recommend a method to test jitter or packetization? Jim Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] And/or periods of large jitter on your network connection. On 3/28/07, Matt [EMAIL PROTECTED] wrote: Could it possibly be a packetization rate issue with your provider? On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] wireless desktop phones
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone.
If they have Cat 3 to each phone, how can it be physically impossible?
Is it *physically* impossible, or is the client emotionally unready for
the implications?
If they are going to VoIP, it is time to do proper cabling and put in
Cat 5e or Cat 6 cable, and do multiple runs per workstation. Diddling
around is for the radio club.
(Another poster pointed out that Cat3 can do 10BaseT, but most Cat3
installations are so old that I wouldn't place my trust in them for
anything requiring the level of reliability people expect of their voice
equipment.)
Voice equipment should work well and all the time -- five 9's
reliability. You're only going to get that by being rigorous.
-Stephen-
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Re: [asterisk-users] Transfering not working - how to debug?
On Wed, 28 Mar 2007, Alan Chandler wrote:
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
these buttons on my handset nothing happens (other than I hear the dtmf
tones on the other end of the line).
roo*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Attended Transfer #2
One Touch Monitor *1
Disconnect Call * *0
I am using the tT options in my dial calls (via a macro)
[macro-extension]
exten = s,1,Dial(${ARG1},20,tT)
I had to fiddle with other things to make this work (needed for the
Siemens CP4600 SIP/DECT phone)
I found that the default timeouts were a bit tight for my likings (and the
people who I was testing this with!)
So in features.conf I have:
transferdigittimeout = 8 ; Number of seconds to wait between digits when
transfering a call
featuredigittimeout = 999 ; Max time (ms) between digits for
; feature activation. Default is 500
[featuremap]
blindxfer = #1; Blind transfer
atxfer = ##; Attended transfer
disconnect = #0; Disconnect
If it's still not working, are you sure the DTMF is being picked
up/transmitted correctly? If it's in-band, is it a codec other than G711?
(which might give you problems)
Gordon
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Re: [asterisk-users] Meetme cant handle more than 5 users in a call?? hmmmm
Matt wrote: Yikes! While I will agree I think Digium needs to do a little better QA (let's not start that war again), this kind of FUD doesn't do anything for the community. I've had Asterisk running with meetme no problem with many more then 5 users. Agreed -- they're treading on dangerous ground. I'm effectively done with Trixbox; it's not any less opaque than Asterisk (if anything, it's moreso) and if I'm going to have to mess around with it, just give me Asterisk. After a while, Fonality's marketing marmelade gets tiresome. “Most people download Asterisk, buy a bunch of phones and then run into a brick wall,” he said. “Those ‘Asterisk rescues’ are a lot of our business right now.” Most people? This guy's got a marketing dude's hand up his shirt. Do you know many businesses that installed their own Norstar or Meridian system? I don't know of *any* business that will dare deploy their own Asterisk -- all the ones I've encountered are using consultants. Now, if he means Most consultants download Asterisk... ;) -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
Re: [asterisk-users] Multi-line phones - Asterisk uses wrong callerid
Do you have multiple devices registering with the 10x extentions? Or is it just the one device? Basically, the phone is not sending the correct Caller-ID, for some reason. Whatever caller-id the phone sends, is what will be sent. On 3/28/07, Drew Gibson [EMAIL PROTECTED] wrote: I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like to resolve:- 1. The person called sees the wrong callerid 2. The CDR records the call against the wrong account 3. Picking up voicemail requires multiple extra steps Is there a way around this?? Scenario:- Phone 1 has three lines 101, 102, 103 Phone 2 has 1 line 202 User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) User 2 at Phone 2 sees call coming from extension 103 instead of 101 With 'sip debug' enabled at the console, I see an INVITE issued (on the Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xx mailbox=101 callerid=User 1 101 sip show peers :- 103/10310.10.10.181 D 5060 OK (157 ms) 102/10210.10.10.181 D 5060 OK (159 ms) 202/20210.10.10.184 D 5060 OK (4 ms) 101/10110.10.10.181 D 5060 OK (160 ms) Asterisk 1.2.15 Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk: recommended installation
On Wed, Mar 28, 2007 at 01:11:05PM -0300, Alejandro Cabrera Obed wrote: Dear all, I'll implement a VoIP system using Asterisk + SIP with softphones; I need to connect LAN and VPN users (about 100-150). What version/installation of asterisk do you recommend tyo me ??? Does [EMAIL PROTECTED] or Trixbox match to my scenario By the way, I use Debian Etch as OS server. apt-get install asterisk Backports of newer versions of Asterisk will hopefully soon also be availble. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] wireless desktop phones
I'm also in the market for a wi-fi phone. My boss currently has a cordless phone and wants to keep the same functionality. We have a robust wireless network in the office and the phone will be staying here, so roaming is not really an issue. Everybody in the office is still going to get wired phones regardless. Matt Gorecki Tempest Technologies http://www.tempesttech.com Stephen Bosch wrote: Jordan Novak wrote: I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? If you enjoy being miserable and having your phones not work, by all means, use a wi-fi phone. Frankly, it's worth the extra coin to get a cable run done to wherever you need it. -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 --- [This E-mail scanned for viruses by Declude Virus provided by Tempest Technologies, LLC] ___ --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] Park No Announce?
Ken Williams wrote: I couldn't find a switch, so I commented line 426 out of res_features.c and recompiled - instant transfer now on Grandstream phones. Below is the line for future reference. ast_say_digits(peer, pu-parkingnum, , peer-language); One of the many, many joys of using open source software! Try that with a binary. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] App_RXFax Problem.
Good day everyone, Hope someone can help me with a spandsp/app_rxfax problem. I've compiled spandsp 0.0.2pre26, and app_rxfax.c from soft-switch.org Both went just fine, and i've checked my libtiff and libxml (along with the devel-s) versions - they're fine. Machine is fedora core 3, x86_64. Asterisk is 1.2.17, zaptel 1.2.15 Dialing in on a zap channel. T1 PRI, esf,b8zs TE110P Card as termination. You can feel free to test call it to see what i'm talking about - 586-408-9849 When i call rxfax(somefile.tif) from an extension, the calling party hears fax tones.. (Beep... Beep... Beep.), but rxfax never goes into 'negotiation mode' where modem style sounds would typically be heard. I've tried calling rxfax from several different fax machines, and the result is always the same. I can hear them 'beeping' at eachother - but they never start negotiating. Below is what shows up in my 'debug' log. Any ideas, Please pass them along. Thanks! John Mar 28 16:15:26 VERBOSE[5868] logger.c: [app_rxfax.so] = (Trivial FAX Receive Application) Mar 28 16:15:45 DEBUG[26457] app_rxfax.c: Got hangup Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC framing OK Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW Changed from phase 2 to 3 Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:31 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:31 DEBUG[26612] app_rxfax.c: FLOW HDLC framing OK Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34
[asterisk-users] SIP OPTIONS dialog not understood
I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a heartbeat of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: -- SIP read from 172.b.c.d:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1 Allow-Events: message-summary Allow-Events: refer Allow-Events: dialog Allow-Events: line-seize Max-Forwards: 70 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=172.b.c.d+1+0+22022a3b CSeq: 445762257 OPTIONS Organization: Supported: 100rel Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060;transport=udp To: sip:[EMAIL PROTECTED] --- (15 headers 0 lines) --- Looking for metaswitch in test (domain 206.b.c.d) Transmitting (no NAT) to 172.b.c.d:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1;received=172.b.c.d From: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=172.b.c.d+1+0+22022a3b To: sip:[EMAIL PROTECTED];tag=as6a59273b Call-ID: [EMAIL PROTECTED] CSeq: 445762257 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:206.b.c.d Accept: application/sdp Content-Length: 0 Is this how OPTIONS is supposed to look? One thing that struck me as curious is that I had to add an extension metaswitch to my test context in my dialplan. Otherwise I got 404's. Can anybody explain (or point to an explanation)? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 and Asterisk
I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --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: Inbound Voice Quality - Speed Change
On Tue, Mar 27, 2007 at 09:30:36PM -0400, Jim Duda wrote: Lacy, I don't have any zaptel cards installed. I do however have ztdummy installed. Is there some tweaks to ztdummy which I might need? Is there a special kernel setting which ztdummy requires? What is the output of zttest ? Run it for a minute or so. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] wireless desktop phones
Hi the list, Think Kirk solution ;-) www.kirktelecom.com This is an DECT/GAP infrastructure solution, and the bases can be seen as something like SIP/DECT gateways. Each wireless phone is like a separate IP phone from Asterisk side. You can use several bases and repeaters (only radio link, no Ethernet cable) to extend the range and have a global coverage into customers buildings. Very incredible, powerfull and scalable solution ! I think it's probably the only one with such a class and commercial grade. Best Regards, Francois BERGERET, France. ___ --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] Cisco 30VIP Phone
On 3/28/07, Jason Parker [EMAIL PROTECTED] wrote: - Derek Whitten [EMAIL PROTECTED] wrote: if i remember right, most of the buttons on those and the 12SP+ phones don't really work because there isn't a button template in * There is a button template, the problem is that most of the softkeys simply aren't implemented. That is the conclusion I came to and was confirmed today in a very brief chat with one of the individuals listed as a developer on the chan_skinny module. He said that they could be implemented. What I would like to know, and do not understand, is the relationship between the code in chan_skinny.c which sets up the softkeys which are implimented and the actual key positions on the phone. With this info, I can hack the code to impliment other of the keys (ie. speed dial, etc.). Thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unsetting Global Vars
How do I clear a global variable for good? I have a situation of needing to use global variables to aide in channel communication, but will be changing the name within a defined scope. Additional Background... I want to get a variable from a channel (child) that is created by another channel (parent), however the execution of the parent channel does not continue until the child channel is gone. So I want to use a global variable as 'scratch' space and later the parent to grab it. Basically I need to be able to do the opposite of variable inheritance. I need to propagate a variable status up the channel chain instead of down. -- Johann Hoehn Project Coordinator, Administration Direct: 270-707-2040 x 4011 Ecommerce Corporation (www.ecommerce.com) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] App_RXFax Problem.
Start with a codec check (sounds like the CNG tone frequencies are out of spec)... MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Wulter Sent: Wednesday, March 28, 2007 4:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] App_RXFax Problem. Good day everyone, Hope someone can help me with a spandsp/app_rxfax problem. I've compiled spandsp 0.0.2pre26, and app_rxfax.c from soft-switch.org Both went just fine, and i've checked my libtiff and libxml (along with the devel-s) versions - they're fine. Machine is fedora core 3, x86_64. Asterisk is 1.2.17, zaptel 1.2.15 Dialing in on a zap channel. T1 PRI, esf,b8zs TE110P Card as termination. You can feel free to test call it to see what i'm talking about - 586-408-9849 When i call rxfax(somefile.tif) from an extension, the calling party hears fax tones.. (Beep... Beep... Beep.), but rxfax never goes into 'negotiation mode' where modem style sounds would typically be heard. I've tried calling rxfax from several different fax machines, and the result is always the same. I can hear them 'beeping' at eachother - but they never start negotiating. Below is what shows up in my 'debug' log. Any ideas, Please pass them along. Thanks! John Mar 28 16:15:26 VERBOSE[5868] logger.c: [app_rxfax.so] = (Trivial FAX Receive Application) Mar 28 16:15:45 DEBUG[26457] app_rxfax.c: Got hangup Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:27 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:29 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC framing OK Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW Changed from phase 2 to 3 Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:30 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:31 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:31 DEBUG[26612] app_rxfax.c: FLOW HDLC framing OK Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:32 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier down Mar 28 16:21:34 DEBUG[26612] app_rxfax.c: FLOW HDLC carrier up Mar 28 16:21:34 DEBUG[26612]
RE: [asterisk-users] Polycom and Asterisk
I would be interested in specifics as I have yet to hear any real issues, a lot of people had some bad taste after 2.0.0, as is to be expected for a first release. I've used 2.0.2, 2.0.3, and now 2.1.0 with Asterisk 1.2 for months without issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett Sent: Wednesday, March 28, 2007 14:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom and Asterisk I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --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: RES: [asterisk-users] Development of new features in Asterisk Manager
On Wed, 2007-03-28 at 15:55 -0300, Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote: Hi Murphy, I am developing an application for integration with Asterisk by Asterisk Manager. When I send a command to asterisk (Example: Action: Originate), many events are raised. I would like to identify what events answer my command. I'm thinking of creating a new property in the events to return the command ActionID. Moacir-- While inserting actionID: headers everywhere is an option, you already can tie these events together using the channel header...? murf Example: I send an Originate: action: Originate actionid: 1234526_PS channel: local/092053469 Exten: 101 Context: default Priority: 1 Async: true Many events are returned: (In this case didn’t answer the call) Event: Newchannel Privilege: call,all Channel: Local/[EMAIL PROTECTED],1 State: Down CallerIDNum: unknown CallerIDName: unknown Uniqueid: 1175078296.23 Event: Newchannel Privilege: call,all Channel: Local/[EMAIL PROTECTED],2 State: Ring CallerIDNum: unknown CallerIDName: unknown Uniqueid: 1175078296.24 . . . Event: Dial Privilege: call,all Source: Local/[EMAIL PROTECTED],2 Destination: SIP/tmaisMG-096eee20 CallerID: unknown CallerIDName: unknown SrcUniqueID: 1175078296.24 DestUniqueID: 1175078296.25 . . . Event: Hangup Privilege: call,all Channel: Local/[EMAIL PROTECTED],2 Uniqueid: 1175078296.24 Cause: 16 Cause-txt: Normal Clearing Event: Hangup Privilege: call,all Channel: Local/[EMAIL PROTECTED],1 Uniqueid: 1175078296.23 Cause: 0 Cause-txt: Unknown Event: OriginateResponse Privilege: call,all ActionID: 1234526_PS Response: Failure Channel: local/092053469 Context: default Exten: 101 Reason: 1 Uniqueid: null CallerID: unknown CallerIDNum: unknown CallerIDName: unknown Only “OriginateResponse” has ActionID. How can I identify the “Newexten, Newchannel, Hangup, Dial, …,” source? Do you understand me? Thanks, []’s Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Steve Murphy Enviada em: quarta-feira, 28 de março de 2007 12:36 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Development of new features in Asterisk Manager On Wed, 2007-03-28 at 11:20 -0300, Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote: Hi, Has anybody customized* anything in Asterisk? * Customized = Development of new features or changes the existent features. I need a new feature in Asterisk Manager and would like to talk about this. This is a good place to discuss it; what you want to do, may already have been done in dozens of places! And people love to brag! :) murf Thanks, Moacir O. de Souza Junior Belo Horizonte - Minas Gerais - Brasil -- Steve Murphy Software Developer Digium 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] Polycom and Asterisk
Matt, I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any problems. What kind of issues did you experience? On 3/28/07, Mike Hammett [EMAIL PROTECTED] wrote: I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. ___ --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 -- Bruce Reeves 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: RE : [asterisk-users] wireless desktop phones
Just be careful with any multi vendor GAP solution (GAP is Generic Access Profile - which means you are supposed to be able to take a handset from any vendor and match it with a base station from any vendor) Basically it's like any standardsure you get basic functionality but you'll often find advanced features are outside the defined spec. 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). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 28 March 2007 5:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [asterisk-users] wireless desktop phones Hi the list, Think Kirk solution ;-) www.kirktelecom.com This is an DECT/GAP infrastructure solution, and the bases can be seen as something like SIP/DECT gateways. Each wireless phone is like a separate IP phone from Asterisk side. You can use several bases and repeaters (only radio link, no Ethernet cable) to extend the range and have a global coverage into customers buildings. Very incredible, powerfull and scalable solution ! I think it's probably the only one with such a class and commercial grade. Best Regards, Francois BERGERET, France. ___ --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] Meetme cant handle more than 5 users in a call?? hmmmm
Dean Collins wrote on 3/28/07 9:27 AM: Meetme cant handle more than 5 users in a call?? H Heh, that's a laugh. We regularly get 40 or more callers in a conference room in MeetMe with no problems. In fact, the call quality is better than some of those 800# conference services we used to use before we had Asterisk. :) The story is likely what hardware you have it running on. If you expect your phone system to be an enterprise-class PBX, it needs to run on enterprise-class hardware, not some leftover 486 box from the back closet. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.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
[asterisk-users] asterisk-addons-1.4 write wrong uniqueid
Evnin' As I didn't find any answer I'll try to rephrase the problem (o; Any idea why the latest asterisk-addons-1.4 write wrong uniqueid into mysql database? Asterisk-1.4.2 creates call record files with the uniqueid prepended: 1175107269-SIP-999-0876c000.wav But into mysql database it writes an uniqueid of: 1175107260.88 but should be: 1175107269 Any idea why the difference? Any why it even writes it in decimal format? cheers rick ___ --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] Dialplan Streaming
Oh poo. No one seems to know. :( Doug Garstang wrote: All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. Doug. ___ --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
