[Asterisk-Users] Uniden UIP200 Review
Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some feedback from an office perspective on the new unit. It is Uniden's first offering into the VoIP market. Main Features which were important to me: Built in 10/100 Switch Speakerphone w/headset port IEEE 802.3af Standard Inline Power (PoE) 2 line 16/char LCD Display 8 Programmable (not soft) Keys QoS [IEEE 802.1 p/q Based and DiffServ G711a/u G729A Codec Support TFTP Auto Configuration Firmware Upgrades (based on mac addressed filenames) The phone also has all the hard buttons you'd expect it to have. Hold, speaker/headset, Volume up and down, Menu, Transfer, Cancel, and Dial (used in lieu of pressing the # key to cut down digit timeouts when on-hook dialing). First, this phone, is relatively inexpensive. I was able to pick one up for $129. Setup and configuration was trying, as the phone ships with absolutely NOTHING in terms of an admin guide. The support areas on the Uniden site were password protected and even the support staff was unaware of all the proper logins and passwords (gotta love supporting new products). Once I gained access to the appropriate admin guide, I whipped up a few of the configuration files on my TFTP server, plugged in the phone and was off and rolling. Or so I thought. There seems to be some minor DHCP issues with the phone currently. It was ignoring my DHCP server's DHCP Offer's and constantly reported DHCP Failed on the LCD. After speaking with a Uniden Developer and sending him an ethereal trace, I hard-coded the IP address to continue my testing. The phone fired up, auto-configured itself via TFTP, and was logged into * in a matter of seconds. Needless to say, at this point, I was extremely pleased to see it actually WORKED. Weak Points: Wimpy Speakerphone: It's extremely easy for the speakerphone itself to over modulate. The microphone however does seem to perform well, even if it is a *little* tin-can'ish. Hold Button: Works as expected, * puts the caller on hold, and they hear MOH. YOU on the other hand hear this really cheesy Nintendo style genre of music locally, produced by the phone. When using speakerphone and placing someone on hold, this is extremely irritating. DTMF: When you have a session, or call active, there is no local DTMF feedback over the handset or speakerphone. While I'm ok with this, I can just picture my entire office on the first day, wondering if they actually pushed the buttons hard enough. So navigating through auto attendant menus can be a little tricky since you're not sure if you actually missed the button, or made solid contact. You can however check the LCD to see if the number you pressed went through. Conclusion: In testing, the phone is an all around solid performer. If they resolve my DHCP issue, I think we probably will go ahead and purchase 20-30 phones to start so that we can get * deployed in the near future. For $130, I don't think I can really complain about the weak points, however I have voiced my opinion on the DTMF and HOLD music to Uniden, so maybe in the near future we'll have some toggles in the TFTP config files make life a little less stressful. Uniden currently has a distributor/wholesaler who will sell to the public. If you're interested in picking up any of these phones to test yourself, the contact information is below. Note: Please keep in mind, Uniden also makes the UIP300 and UIP312. These phones *only* support H323. The UIP400 is the equivalent model of the 300, but will support SIP and is currently in development. Contact: Aimee @ Teledynamics (800) 847-5629 ext.110 or, [EMAIL PROTECTED] Brian D'Arcy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 729 licence on scsi
* Mark Spencer; [EMAIL PROTECTED] on 07 May, 2004 wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. In addition to eliminating the crappy copy protection and what will happen if the box has more than one ethernet card -- Togan Muftuoglu ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail: upgraded?
I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] 729 licence on scsi
On Fri, 2004-05-07 at 23:00, Mark Spencer wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. **smootch** (I won't even ask you how long 'a little while' is either ;-) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Concept for line appearances and bridging: anyone?
OK, here's a configuration challenge: I want to have certain line appearances able to be interrupted by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all ring on line appearance 1234 (there is a specific line labelled 1234 on each phone) 3) User A picks up the ringing call on 1234. Line X and User A are bridged. 4) User B saw the caller ID on the call before it was picked up by user A, but she wants to talk to the caller as well since she has some relevant information. User B picks up the phone and pushes the 1234 extension button. A warning tone is played into the conversation between X and User A, and then User B is bridged into the conversation. User B then talks with X and User A, and then hangs up. This is _extremely_ relevant to office PBX systems. In fact, it's one of the most used features - the ability to share a call with other people in the office just by hitting the right line appearance button. Has anyone come up with a reasonable solution to delivering this feature? For small offices, this is really a mandatory feature though as the number of calls increases this becomes more useless in an inbound setting (though as a workgroup feature it gains usefulness with size of the organization. I'll skip the business cases for why this is a good idea and leave it as an exercise for the reader.) I have come up with ideas on doing this with some really horrible, nasty, awful ideas that involve MeetMe rooms, but shudder... they're really not the right way to do it. There must be some clever way of doing this with a new channel specification that would allow bridging into an existing channel identifier. I.E.: Dial(Bridge/SIP/2203-bed5) Other related topics: - The auto-dial I can handle with PLAR (hotline calling - pick up the phone, and automatically a number is dialed) and DISA on the Asterisk side. In other words, when someone picks up line #1 on their Cisco 7960 (or whatever phone) I can have the system auto-dial into my * server. Using the caller ID, I can determine what line they're calling from. If there is nobody on that line appearance, then I can give them a DISA to allow them to dial a regular call, as if the auto-ringdown didn't happen. - This feature becomes useful now that we have some phones that support SUBSCRIBE methods to allow other phones to show who is on what lines. We can _see_ who is on the line, but there is no ability to add other lines to the call without transferring to a MeetMe (which then causes call control to be lost, and is a hassle, etc. etc. etc.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI, multi D channels and conventional PBXs
try setting immediate=no for that span Jason At 18:12 07/05/2004 +0100, you wrote: Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put the PBX onto a span 2, it triggers the D-channel and all seems hunky dory - until you try to acquire a line from * - this gives me: -- Extension '' in context 'blah' from '' does not exist. Rejecting call on channel 6, span 2 Any suggestions most welcome! Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: PRI, multi D channels and conventional PBXs (brian)
Hi bkw Yep, which is going to be a huge problem since it's only taking a line and not doing any transmittal until after you get a line out, the line of course is being rejected before I can even get there :( Of course I can't even establish connectivity to the telco whilst having it peered to the PBX too due to the D channel issue :( Lee From: brian [EMAIL PROTECTED] -- Extension '' in context 'blah' from '' does not exist. Rejecting call on channel 6, span 2 Looks like the pbx isn't sending any info such as called exten bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One-way SIP question
Ok, here's a good one -- I've tried a lot of different things, searched the archives, etc. I've signed up with a VOIP provider and coaxed the SIP settings out of them. I have username (=phone), password, sip.provider.com and proxy.provider.com. Using these settings, in Asterisk or X-lite, I can place outgoing calls, but do not receive incoming calls. With the same settings on a Sipura SPA-2000 I can receive incoming calls, but not make outgoing calls. Any easy place to look? Sipura and X-Lite both run behind a NAT firewall. The Asterisk server is also behind the firewall, but on a static-route (I have five static IP addresses, * occupies one of those, and all non-static machines -- 2 Vonage ATAs, the Sipura, several computers -- all NAT out on one address) Should I syslog the sipura to see what's wrong with it? Not sure I'll be able to read those logs. Thanks -- J ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MPG123 errors
When I put someone on hold audio doesnt play and i get mpg123: unknown option "mono", Any ideas. I searched wifi and archives. Kyle
RE: [Asterisk-Users] Cisco 7940 Phones as paging system?
Got so many people asking for it, heres what I used for the intercom announce: http://www.jsci.net/asterisk/intercom-tone.gsm Its not great, but it does the job. Actually trying to find something better -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Antkowiak Sent: Friday, May 07, 2004 4:30 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? This is what we have for this customer. They have five phones right now. Their normal extensions are 610x, but for intercom its 510x: exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten = 5101,2,Congestion If you want the wav file, let me know. If you make your own, be sure to put a 1-2 second pause in the beginning, because when the cisco answers it takes a second or to before it will send any audio to the speaker. -Original Message- From: mitchel [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? Hey Joe, Could I get a sample config for playing some intro tones on the intercom? I have the same thing but nobody is using it now because they are afraid of having someone call in and listen in so we need some way to announce the incoming intercom call. Thanks, Mitchel Joe Antkowiak wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Works great, customer is happy. As far as an intro-tone, you can set the dial command to play a sound (using the announce option) before the call is connected. I grabbed a simple tone wav file, and made it play that. Now, when the intercom ext is called, it plays the tone on the destination phone, and wa-la, intercom So it works. Let me know if you need sample configs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, May 07, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system? Hi! able to support intercom/paging. Having searched the archives, it appears that this question was asked about 6 months ago, and the answer was that the Cisco phones support this using SCCP and having one line set to auto-answer, but at the time this was not supported in the SIP image. Is this still the case? Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (- echo for the remote speaker talker, and too thin sound from the person in the room). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] 729 licence on scsi
Excellent :) That was the clunkiest thing I have ever seen. Looking forward to your beta code. Also, please make sure that users switching from the old binary to the new binary does not lose their licenseI remember reading something about only being able to register 3x. On Fri, 7 May 2004 16:00:33 -0500 (CDT), Mark Spencer [EMAIL PROTECTED] wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. In addition to eliminating the crappy copy protection code, the new version is approximately twice as fast as the VoiceAge code, which DOES NOT MEAN YOU WILL BE ABLE TO DOUBLE THE NUMBER OF CHANNELS PER BOX but does mean that you should be able to get substantially more channels per box. Anyway I'll post again on here when we're ready for beta testing, and anyone that has bought a license for the voiceage code will get to upgrade to the new code free of charge, of course. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WI FI IP phones??
John, Check out www.vocera.com instead then. Built for this exact situation. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Saturday, 8 May 2004 2:27 AM To: Asterisk Subject: Re: [Asterisk-Users] WI FI IP phones?? No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WI FI IP phones??
I guess vocera doesn't have any RF engineers to tell them they can't do it. Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Musone Sent: Friday, May 07, 2004 9:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] WI FI IP phones?? Why not vocera? http://www.vocera.com they seem to have the exact product you are looking for and seem to primarily server hospitals.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Friday, May 07, 2004 1:06 PM To: Asterisk Subject: Re: [Asterisk-Users] WI FI IP phones?? Hmm I'll look into it. Thanks. On Fri, 2004-05-07 at 12:54, John Fraizer wrote: James Moran wrote: No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John Um, I'm not so sure that you're going to be able to run WiFi at a hospital. The life safety/support equipment is most likely not certified to be resistant to 2.4Ghz interference. It's been a while since I looked up ISM allocations but, I can tell you that I've seen many good ideas shot down because of the potential to interfere with the medical equipment. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
Joe Antkowiak wrote: exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten = 5101,2,Congestion That's not too bad, but how do you page a group of phones...like a real intercom? That's what I'm dying to know! -- /* Ian A. Underwood - [EMAIL PROTECTED] - http://www.agentgreen.org There are 4 boxes to use in the defense of liberty: soap, ballot, jury, ammo. Use in that order. Starting now. */ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 729 licence on scsi
Mark Spencer wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. In addition to eliminating the crappy copy protection code, the new version is approximately twice as fast as the VoiceAge code, which DOES NOT MEAN YOU WILL BE ABLE TO DOUBLE THE NUMBER OF CHANNELS PER BOX but does mean that you should be able to get substantially more channels per box. Anyway I'll post again on here when we're ready for beta testing, and anyone that has bought a license for the voiceage code will get to upgrade to the new code free of charge, of course. Mark Hmm, which code is used for the new h.729 Codec. And which license. Here, in my NOT FREE, Ex Communist country is completely legal to have GPL-ed g.729 code. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 729 licence on scsi
At 4:00 PM -0500 on 5/7/04, Mark Spencer wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. In addition to eliminating the crappy copy protection code, the new version is approximately twice as fast as the VoiceAge code, which DOES NOT MEAN YOU WILL BE ABLE TO DOUBLE THE NUMBER OF CHANNELS PER BOX but does mean that you should be able to get substantially more channels per box. Anyway I'll post again on here when we're ready for beta testing, and anyone that has bought a license for the voiceage code will get to upgrade to the new code free of charge, of course. Mark WOO HOOO!! Down with G.729! Long live G.729! JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callwaiting callerid on 390s?
Anyone got callwaitingcallerid working succesfull on nortel/aastra/.../... 390 ADSI Phone? It will be great if someone share some ADSI Scripts for these phones also. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P keeping PSTN line Offhook
Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WI FI IP phones??
Not sure if my other message got through. Wifi limitations with voip are a function of # of concurrent active calls per access point (in addtion to which codecs used). A single floor of the hospital might have many many access points. If you just need a way to contact nurses on call, my guess is you would never have all 30 phones active at once. This would be in your favor, since just having a phone on and in standby is not going to eat up much bandwidth. Howeve since peoples lives may be on the line the network would need to be really well engineered and coverage would need to be designed with concurrent calls in mind rather than the usual testing for signal strength. In addition to the generel active calls/AP issue there are protocols like SVP, Spectra Link Protocol, that provide for some type of quality of service for Wifi. I don't really recomend Spectra link, mainly because they wanted a $5000 commitment just to demo their product. There is a qos spec for qos on wifi in the works, this won't magically make you able to have more calls per AP, but it might help calls in session from being starved out. I am pretty sure that what you want to do can be done at least with the Spectra stuff, since I have talked to tech director of a new middle school that is using this for their phone system. Every classroom teacher has one of these and I think most of the other phone users in the building. Another little trick floating around is for the Linksys APs. Again not necessarily recommending but there are custom firmware images (since they run Linux) that provide QOS, I think there may even be an image with a builtin SIP proxy. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting James Moran [EMAIL PROTECTED]: No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MPG123 errors
Use mpg123 version 0.59r On Fri, 2004-05-07 at 17:57, Kyle Hagan wrote: When I put someone on hold audio doesnt play and i get mpg123: unknown option mono, Any ideas. I searched wifi and archives. Kyle -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WI FI IP phones??
Nope I works differently, they use it in a few hospitals here in Sydney, basically it works like the new gsm 'push to talk' service being rolled out, basically limited number of frequencies, voice 'envelope' being delivered as a best case availability basis. It's not a 'held up' tdma style call flow that wifi phones are. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Saturday, 8 May 2004 9:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] WI FI IP phones?? I guess vocera doesn't have any RF engineers to tell them they can't do it. Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Musone Sent: Friday, May 07, 2004 9:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] WI FI IP phones?? Why not vocera? http://www.vocera.com they seem to have the exact product you are looking for and seem to primarily server hospitals.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Friday, May 07, 2004 1:06 PM To: Asterisk Subject: Re: [Asterisk-Users] WI FI IP phones?? Hmm I'll look into it. Thanks. On Fri, 2004-05-07 at 12:54, John Fraizer wrote: James Moran wrote: No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John Um, I'm not so sure that you're going to be able to run WiFi at a hospital. The life safety/support equipment is most likely not certified to be resistant to 2.4Ghz interference. It's been a while since I looked up ISM allocations but, I can tell you that I've seen many good ideas shot down because of the potential to interfere with the medical equipment. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
This isn't really the issue. Up until a week ago or so everything worked fine with a hallf duplex hub. Now it doesn't - so I suspect some code change made in * is responsible. I think * must maintain backwards compatibility with existing hardware or many people will get fed up with constant degradation of sound quality. Iain --On Friday, May 07, 2004 14:15:47 -0600 James Sizemore [EMAIL PROTECTED] wrote: I checked-out CVS Head today to get realm support, I have over hundred Cisco phone on my servers and I have not noticed any Qos problems. You may want to check the duplex of your switches and Asterisk boxes. If you don't have full duplex, that is more then likely your problem. Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Indication Busy to a ZAP ISDN channel
Hi,
I am stuck with my extensions.conf and would appreciate a small hint from the
ISDN experts.
What is the correct way to indicate a busy condition to a calling ISDN zap
channel (TE410P) when a local SIP ext. is busy?
I have
[pstn-in]
exten = *591,1,Dial(SIP/${EXTEN},45,r)
exten = *591,102,Busy
and get
-- Executing Dial(Zap/1-1, SIP/*591|45|r) in new stack
-- Accepting call from '*0420' to '*591' on channel 1, span 1
-- Called *591
-- Got SIP response 486 busy back from ***.***.***.222
-- SIP/*591-ffeb is busy
== Everyone is busy at this time
-- Executing Busy(Zap/1-1, ) in new stack
*CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
Zap/1-1 (pstn-in*591 102 ) Ringing Busy (Empty)
if the SIP extension is busy. The calling ISDN party hears ringtone then instead
of busy indication.
Asterisk seems to know the ext. is busy but doesn't do the correct signalling in
the zap ISDN D-channel.
What is the correct way to do this, of course without answering the channel and
thus producing costs to the caller?
Thanks and regards,
Jan Baumann
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[Asterisk-Users] authorise with h323 client at the * via gatekeeper
Hi folks, I am using opengk to handle h323 calls. * and my clients register at opengk successfully. But everyone can register to my gk?? Is there a way to restrict the clients by using the authorisation of h323.conf ?? Cheers, Harald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 729 licence on scsi
if i have ordered one lic. and now i have realized i need two lic for one call (2 cannels one to provider one to sipphone) can i install 2. lic with another reg code ? nico Mark Elkins wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? Who do I talk to now? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 729 licence on scsi
if i have ordered one lic. and now i have realized i need two lic for one call (2 cannels one to provider one to sipphone) can i install 2. lic with another reg code ? You shouldn't need two if the SIP phone and the provider are both using g.729 so long as you dont' expect Asterisk to see the audio path (i.e. hear DTMF, etc.) -- Asterisk can bridge g.729 calls without a license in this case. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Concept for line appearances and bridging: anyone?
John, i think MGCP has this feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Friday, May 07, 2004 5:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Concept for line appearances and bridging: anyone? OK, here's a configuration challenge: I want to have certain line appearances able to be interrupted by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all ring on line appearance 1234 (there is a specific line labelled 1234 on each phone) 3) User A picks up the ringing call on 1234. Line X and User A are bridged. 4) User B saw the caller ID on the call before it was picked up by user A, but she wants to talk to the caller as well since she has some relevant information. User B picks up the phone and pushes the 1234 extension button. A warning tone is played into the conversation between X and User A, and then User B is bridged into the conversation. User B then talks with X and User A, and then hangs up. This is _extremely_ relevant to office PBX systems. In fact, it's one of the most used features - the ability to share a call with other people in the office just by hitting the right line appearance button. Has anyone come up with a reasonable solution to delivering this feature? For small offices, this is really a mandatory feature though as the number of calls increases this becomes more useless in an inbound setting (though as a workgroup feature it gains usefulness with size of the organization. I'll skip the business cases for why this is a good idea and leave it as an exercise for the reader.) I have come up with ideas on doing this with some really horrible, nasty, awful ideas that involve MeetMe rooms, but shudder... they're really not the right way to do it. There must be some clever way of doing this with a new channel specification that would allow bridging into an existing channel identifier. I.E.: Dial(Bridge/SIP/2203-bed5) Other related topics: - The auto-dial I can handle with PLAR (hotline calling - pick up the phone, and automatically a number is dialed) and DISA on the Asterisk side. In other words, when someone picks up line #1 on their Cisco 7960 (or whatever phone) I can have the system auto-dial into my * server. Using the caller ID, I can determine what line they're calling from. If there is nobody on that line appearance, then I can give them a DISA to allow them to dial a regular call, as if the auto-ringdown didn't happen. - This feature becomes useful now that we have some phones that support SUBSCRIBE methods to allow other phones to show who is on what lines. We can _see_ who is on the line, but there is no ability to add other lines to the call without transferring to a MeetMe (which then causes call control to be lost, and is a hassle, etc. etc. etc.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P keeping PSTN line Offhook
Shahid wrote: Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 I have the exact same thing happening. I was able to track it down to the Music On Hold. Scenario: - My boss calls my cellphone - I do not pick up - My cellphone's voicemail picks up - He hangs up - Asterisk plays On Hold music to voicemail until voicemail on cellphone times out - Zap channel is stuck -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 729 licence on scsi
Togan Muftuoglu wrote: and what will happen if the box has more than one ethernet card Mark is smarter than Voiceagehe will make it work. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
On Fri, 7 May 2004, Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). I had jittery audio with dropouts on a 7960 with SCCP, and started testing SIP hoping it would be better (based on the reports of the SIP-to-IAX2 timestamping issue). Here's my experience: * As Brian mentions, when the other end of the call is from a non-VoIP path (e.g. Zaptel interface) the audio is fine. * Calls over IAX start out okay, but within a few seconds the audio starts jittering. It gets progressively worse until about a minute into the call (often less), by which time audio is unintelligible. Calling the same number over the same IAX connection from an analogue phone attached to a SIP-image ATA-186 which in turn is plugged into the PC port of the same 7960 gives perfect audio. * Calls over SIP are stable; I had an intermittent problem where audio into the 7960 would stop completely for up to three seconds, but that seems to be gone after doing a CVS update. Side note: when I had this audio dropout problem, making the same call without * in the audio path (by using canreinvite=yes and removing t and T from Dial) resulted in perfect audio. I'll try someone's suggestion to disable the jitterbuffer to fix the IAX2 problem, but I thought that the jitterbuffer was supposed to help this kind of problem... Besides, the same call over an ATA or using X-lite is perfect. Before anyone jumps in, yes, as soon as I can get there I will hit the bug tracker. Cheers, Vic Cross PS: I know that folks generally dislike 'me too' messages, but this time Too Bad -- I'm trying to provide more info to help anyone that might be working on problems. rant I hope that Iain was exaggerating when he described his bug-reporting experience. Many * users are unable to commit the time to poring over hundreds of lines of uncommented C code and ethereal traces with thousands of packets captured. So, as our way of trying to help, we provide e-mails like this either in response to or as an attempt to gather more information about the problem. To try and get people talking about a problem. How is does it help to jump on someone who is trying to get resolution to a problem -- by driving them toward OpenPBX or VOCAL? A few former colleagues of mine may soon be about to learn (unfortunately) that you can only piss off a customer so many times. To the Asterisk developers, bug marshals, and coders: I am jealous of you! You've created a wonderful thing. I'd love to be able to spend the amount of time I'd like to on Asterisk. I'd love to be able to do more to fix bugs and develop features. But I can't. Don't think less of me because of that. /rant VC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 729 licence on scsi
Jeremy McNamara wrote: Togan Muftuoglu wrote: and what will happen if the box has more than one ethernet card Mark is smarter than Voiceagehe will make it work. Jeremy McNamara That isn't saying much. The village idiot is smarter than VoiceAge. :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P keeping PSTN line Offhook
You might try callprogress=no It sort of sounds like noise (or analog phones) on the pstn side are signaling the x100p to go off hook and possibly do other things. Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
On Fri, 7 May 2004, Ian A. Underwood wrote:
Joe Antkowiak wrote:
exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten = 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real
intercom? That's what I'm dying to know!
in extensions.conf:
[globals]
INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...
Then the extension is as per Joe's example, but replacing SIP/5101 with
${INTERCOMLINES}.
Extending this, you could set up various intercom numbers for different
parts of the office...
[globals]
SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
MKTGINTERCOM=SIP/Marketing1-6...
...
[yourcontext]
exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
...
exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))
Cheers,
Vic Cross
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Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) and then we would have a true paging
system..
- Original Message -
From: Vic Cross [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 08, 2004 10:11 AM
Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
On Fri, 7 May 2004, Ian A. Underwood wrote:
Joe Antkowiak wrote:
exten = 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten = 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real
intercom? That's what I'm dying to know!
in extensions.conf:
[globals]
INTERCOMLINES=SIP/Alice6SIP/Bob6SIP/Chuck6...
Then the extension is as per Joe's example, but replacing SIP/5101 with
${INTERCOMLINES}.
Extending this, you could set up various intercom numbers for different
parts of the office...
[globals]
SALESINTERCOM=SIP/Sales1-6SIP/Sales2-6...
MKTGINTERCOM=SIP/Marketing1-6...
...
[yourcontext]
exten = 5101,1,Dial(${SALESINTERCOM},10,tA(tone))
exten = 5102,1,Dial(${MKTGINTERCOM},10,tA(tone))
...
exten = 5110,1,Dial(${SALESINTERCOM}${MKTGINTERCOM}${...},10,tA(tone))
Cheers,
Vic Cross
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[Asterisk-Users] need working loopstart config - t100p
I am connecting a t100p to a b8zs, superframe, loopstart t1. Previously I've attached to em wink and pri lines with no problems; however I seem to be missing something. Should it be fxols in the zaptel.conf (smartjack to x100p) or fxsls? With em wink the dnis set was 100, so it was easy to make an extension 100; and it would answer incoming calls. How do you get a loopstart to answer incoming calls? Thanks, t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
Vic, The problem you're having has been discussed multiple times on this list, and can be easily seen using ethereal to inspect the timestamps contained within the rtp packets sent to the 7960 phone. There are several issues involved, including: 1. the cisco phones drop any rtp packet that is not exactly 160 milliseconds between successive packets (thus causing choppy audio). That drop function seems to be the result of cisco changing DSPs in their v6.x code. I've not heard of anyone running v5.x sip code with the problem. 2. iax2 had a bug in it that Mark fixed last month. The bug resulted in iax2/gsm timestamps that were erratic when they should have been exactly 20 milliseconds between successive packets. 3. Code was added to rtp.c about a month or so ago that ties the iax2/gsm timestamps directly to the sip/rtp timestamps. When that code was added, it made the iax2 erratic timestamps and Cisco's dropping of packets extemely obvious to iax2 users. Other non-iax2 users are not impacted by this. Cisco phones seem to be the only ones impacted by this. There are three short term fixes available to you: a. upgrade (or insist your service provider) upgrade their iax2 code. (I don't believe the Stable branch has the fix in it as yet.) NuFone and some others have done that a few weeks ago. b. remove the two or three lines that were added in rtp.c (although Mark is discouraging this approach for other reasons), or, go back to source code cvs from about early March. c. Change the 7960's from v6.x code to v5.x code (and open a trouble ticket with Cisco). I've not heard anyone suggest that dropping rtp packets with uneven timestamps is necessary, a standard, or anything else. Therefore believe it's an anomaly that crept in with the DSP change in the sip v6.x code from Cisco. Rich On Fri, 7 May 2004, Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). I had jittery audio with dropouts on a 7960 with SCCP, and started testing SIP hoping it would be better (based on the reports of the SIP-to-IAX2 timestamping issue). Here's my experience: * As Brian mentions, when the other end of the call is from a non-VoIP path (e.g. Zaptel interface) the audio is fine. * Calls over IAX start out okay, but within a few seconds the audio starts jittering. It gets progressively worse until about a minute into the call (often less), by which time audio is unintelligible. Calling the same number over the same IAX connection from an analogue phone attached to a SIP-image ATA-186 which in turn is plugged into the PC port of the same 7960 gives perfect audio. * Calls over SIP are stable; I had an intermittent problem where audio into the 7960 would stop completely for up to three seconds, but that seems to be gone after doing a CVS update. Side note: when I had this audio dropout problem, making the same call without * in the audio path (by using canreinvite=yes and removing t and T from Dial) resulted in perfect audio. I'll try someone's suggestion to disable the jitterbuffer to fix the IAX2 problem, but I thought that the jitterbuffer was supposed to help this kind of problem... Besides, the same call over an ATA or using X-lite is perfect. Before anyone jumps in, yes, as soon as I can get there I will hit the bug tracker. Cheers, Vic Cross PS: I know that folks generally dislike 'me too' messages, but this time Too Bad -- I'm trying to provide more info to help anyone that might be working on problems. rant I hope that Iain was exaggerating when he described his bug-reporting experience. Many * users are unable to commit the time to poring over hundreds of lines of uncommented C code and ethereal traces with thousands of packets captured. So, as our way of trying to help, we provide e-mails like this either in response to or as an attempt to gather more information about the problem. To try and get people talking about a problem. How is does it help to jump on someone who is trying to get resolution to a problem -- by driving them toward OpenPBX or VOCAL? A few former colleagues of mine may soon be about to learn (unfortunately) that you can only piss off a customer so many times. To the Asterisk developers, bug marshals, and coders: I am jealous of you! You've created a wonderful thing. I'd love to be able to spend the amount of time I'd like to on Asterisk. I'd love to be able to do more to fix bugs and develop features. But I can't. Don't think less
RE: [Asterisk-Users] sip notify from iconnect
Just a quick FYI.. I now only use iconnecthere for incoming calls, I am phasing that out. If a company doesnt want to give us the information to properly use their services, then they dont need my money. I am now using voice pulse and I love it. I just wish they would have more exchanges. Have a good day!! Zac From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sathya Sent: Friday, April 30, 2004 10:55 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] sip notify from iconnect Thanks, Zac. yes I am connecting to sipauth.deltathree.com. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Zac Amsler Sent: Friday, April 30, 2004 12:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] sip notify from iconnect I would guess that is a keep alive. Are you connection to sipauth.deltathree.com? Zac On Fri, 2004-04-30 at 13:34, Sathya wrote: Hello, Recently I am seeing this message on my asterisk console received fromIconnect. Apr 30 11:37:21 NOTICE[1125329600]: chan_sip.c:5648 handle_request: Unknown SIP command 'NOTIFY' from '213.137.73.41' It is prety annoying as it appears once every four seconds. I've seensimilar posts inthe archives which points me to NAT keep alives being send by the remote end. I am actually on public internet and there is no NAT involved in, hence no nat config in sip.conf. Why would iconnect send notify messages ? Is there a config setting that I could make so that this message not being sent to me. Cheers SW
[Asterisk-Users] Routing by Called interface
Hey everyone, I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Thanks! Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AVM B1 ISDN Call forwarding
Hi, i want forward a call witch is comming over isdn (avm b1 witch i have) out to isdn (same card 2. b channel). The call is comming (one b channel open one is free) the forwarding is processed (snom 200) all seems correctly. Then the message that the b channels all busy, but so is it not. Forwarding to a sip phone works. Can anyone help me with that ? nicolas SNIPS: -- == DISCONNECT_IND PLCI=0x201 REASON=0x34a2 -- CAPI[contr1/outgoingmsn]/78 is busy -- CAPI Hangingup -- removed pipe for PLCI = 0x201 == Everyone is busy at this time -- Executing Congestion(Local/nummer to forward@default-f1df,2, ) in new stack -- Local/nummer to forward@default-f1df,1 is circuit-busy == Spawn extension (default, nummer to forward, 2) exited non-zero on 'Local/nummer to forward@default-f1df,2' == Everyone is busy at this time -- Executing DigitTimeout(CAPI[contr1/outgoingmsn]/77, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(CAPI[contr1/outgoingmsn]/77, 10) in new stack -- Set Response Timeout to 10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
At 10:31 AM -0400 5/8/04, Billy Huddleston wrote: That won't work.. That'll DIAL multiple phones/extensions, but will only bridge 1 of them when it auto-answers.. What we need is a way to have something like meetme call multiple extensions and bridge them to a meetme confrence (all of them muted but the admin of course, as it's a one way page) and then we would have a true paging system.. OK, I typically would badger people into looking in Google for this, but I'll be darned if I can't find this post on Google myself (search for Office-wide paging with Asterisk or AGI(callall) so I'll re-post here. This is a terrible hack. Someone _please_ make this cleaner. I'm looking at how to add this to the Wiki, but I don't see anything that's obviously marked as start new thread or similar links. If anyone is feeling ambitious, please add the stuff below. JT Date: Sun, 18 Jan 2004 17:22:11 -0700 To: asterisk-users-lists.digium.com From: John Todd [EMAIL PROTECTED] Subject: Office-wide paging with Asterisk and Cisco 7960 7940 phones I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the configuration into sip.conf as you normally would. I figure you have enough clue to create a new line in sip.conf and on your Cisco phones at this point. Go into settings - Call Preferences - Auto Answer (intercom) and then make the new line you've just created as auto-answer. (I wish there was a way to do this via the configuration file! Having to set this while sitting in front of the phone is silly and wasteful.) Now that you have created a valid Asterisk-capable SIP line that auto-answers, here's how you get the paging features to work: Here's what I have in extensions.conf: [conference] exten = ,1,AbsoluteTimeout(21) exten = ,2,AGI(callall) exten = ,3,MeetMe(,dq) exten = ,4,Hangup exten = t,1,Hangup exten = T,1,Hangup exten = h,1,Hangup ; [add-to-conference] exten = start,1,AbsoluteTimeout(20) exten = start,2,MeetMe(,dmq) exten = h,1,Hangup exten = t,1,Hangup exten = T,1,Hangup Here are the contents of /var/lib/asterisk/agi-bin/callall #!/bin/sh cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing Make sure to make the script executable. And then for every extension I have as an auto-answer, I have a file like this in /var/lib/asterisk/agi-bin : Channel: SIP/2006 Context: add-to-conference Extension: start Priority: 1 CallerID: Office Pager So, I have three lines that are configured for automatic answering - SIP/2006, SIP/2007, SIP/2008. I have three files named 2006-conf, 2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into the outgoing call spool directory every time I call extension . These three lines are the auto-answer lines on each of the three phone devices I'm experimenting with. Now, dial from any phone and you should have one-way paging. Voila! People who use the pager may have to get used to waiting 1-2 seconds before speaking to allow all the phones to catch up with the audio stream. All of the phones hang up after 20 seconds, regardless of if the person originating the page has stopped talking. Change the AbsoluteTimeout values to increase this interval. If you want a really confusing loud mess, then change the dmq options to dq and you'll get an N-way conversation going with everyone who has a phone. Bad. If you want a really interesting office surveillance tool, change the dmq to dt and you'll suddenly be listening to all of the extensions in the office, like some kind of mega-snoop tool. Useful for after-hours listening throughout the entire office. Someone should improve my scripts with the following changes: 1) AGI should automatically show the caller ID of the person originating the call instead of a generic pager address 2) The AGI should take arguments of what extensions to call and then dynamically create the list of files that get copied out to the /var/spool/asterisk/outgoing directory JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Concept for line appearances and bridging: anyone?
At 9:27 AM -0400 on 5/8/04, Todd Lieberman wrote: John, i think MGCP has this feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Friday, May 07, 2004 5:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Concept for line appearances and bridging: anyone? OK, here's a configuration challenge: I want to have certain line appearances able to be interrupted by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: [snip] MGCP may have this feature, but Asterisk should be able to provide this functionality on any channel type, not just MGCP. The fact that we can manipulate the audio on the server with * implies that we can mix any two channels in an arbitrary way. This implies (of course) that we keep the audio channel going through our * server, but for most PBX environments this isn't a concern, and in fact is a desirable goal. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List of online sip users
Hello list, is it possible to see all online users? I have configure a isdn2sip gateway in the company I work. Now, the question: Is it possible to show all colleague which people where reachable with this gateway? greetings and thanks, Holger _ Der WEB.DE Virenschutz schuetzt Ihr Postfach vor dem Wurm Netsky.A-P! Kostenfrei fuer alle FreeMail Nutzer. http://f.web.de/?mc=021157 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Routing by Called interface
On Sat, 2004-05-08 at 10:52, Chris Wilson wrote: I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Yup! Check your trash folder. This was discussed on this list in the past 7 days. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List of online sip users
is it possible to see all online users? I have configure a isdn2sip gateway in the company I work. Now, the question: Is it possible to show all colleague which people where reachable with this gateway? Have a look at Monastery http://www.unslept.com/monastery/ and/or http://graphics.cs.uni-sb.de/VoIP/devel/ -- http://graphics.cs.uni-sb.de/VoIP/ pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Routing by Called interface
On Sat, 2004-05-08 at 10:52, Chris Wilson wrote: I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Yup! Check your trash folder. This was discussed on this list in the past 7 days. I didn't get this yet. The helicopter noise still sounds I'm changing the cables today to eliminate any interference possibility. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfering with Grandstream Phones
Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM II and Siptone phone on eBay
Sorry to post this here also, but the biz list doesn't seem to have much traffic yet. I have a brand new SNOM 200 IP phone and also a new Siptone II phone available on eBay, see http://tinyurl.com/2pbng They are surplus after a customer cancelled an order. Please direct all followup questions or bids on eBay, not here. Thanks. -Ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfering with Grandstream Phones
I have the same problem with 2 ATA-286s, DTMFMODE=info and Dial command with Tt options. Only the caller is able to transfer the call with the # key. The callee is not able to transfer the call using # key, unless the codec is ULAW and the DTMFMODE is inband. I suspect the problem is due the GS unit because I failed to detect any DTMF INFO packets going into asterisk from the callee using ethereal. DTMF INFO packets were detected from the caller, though. MPlus - Original Message - From: Paul Tyreman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 09, 2004 2:09 AM Subject: [Asterisk-Users] Transfering with Grandstream Phones Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfering with Grandstream Phones
On 8-May-04, at 12:09 PM, Paul Tyreman wrote: Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have the same problem (attempting to transfer a call made to my BT102 will result in that call being disconnected/hung). Workaround is to use '#' to transfer instead of the 'transfer' button on the phone. RC I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List of online sip users
Holger, From the Asterisk CLI, type: sip show peers This will show you all users currently registered with Asterisk. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Holger Zimmermann Sent: Saturday, May 08, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] List of online sip users Hello list, is it possible to see all online users? I have configure a isdn2sip gateway in the company I work. Now, the question: Is it possible to show all colleague which people where reachable with this gateway? greetings and thanks, Holger _ Der WEB.DE Virenschutz schuetzt Ihr Postfach vor dem Wurm Netsky.A-P! Kostenfrei fuer alle FreeMail Nutzer. http://f.web.de/?mc=021157 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List of online sip users
In cvs head version of chan_sip, there's two new CLI commands: * sip show peer name Show details of peer name - configuration, registration status etc * sip show subscriptions List active SIP subscriptions to extension state changes in Asterisk /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List of online sip users
Can you tell me if the md5secret stuff is broken? I noticed lastnight I went by the wiki instructions and it didnt work. Alos if you change from secret to md5secret then reload and do a sip show peer XXX it will say both are set. bkw - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 08, 2004 12:53 PM Subject: Re: [Asterisk-Users] List of online sip users In cvs head version of chan_sip, there's two new CLI commands: * sip show peer name Show details of peer name - configuration, registration status etc * sip show subscriptions List active SIP subscriptions to extension state changes in Asterisk /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover Scenario - synchronizing voicemail key files
I currently have several asterisk servers geographically distributed (for automatic fail-over in the event of either a network or server problem). My carrier delivers to each server based on the same priorities that I have set inthe DNS SRV records which the clients point to. Users always have dialtone regardless of a single server failure. In addition, once they have re-registered with the backup server they will receive calls as normal until the primary becomes available again. However, there is a slight issue in a few circumstances (and others not listed): (1)The carrier can see the primary, but the clients can't (or vice versa) meaning that a user won't receive the call (butcan make calls, andinboundwill go into voicemail). (2)A user receives a voicemail while temporarily failed-over to another server, then re-registers with the primary server (meaning the voicemail will be "stuck" on the secondary server). The solutionI amusingis to run rsync to synchronize configuration files, and (hopefully soon)voicemail between all of my servers. The config files are no problem since changes will always be made on the primary, then propagated out (I run a script to look at the checksum of /etc/asterisk every 5 minutes, if it is different i execute a reload). However, the voicemail (and potentially sound files, etc.) are a different story. Best I can tell, voicemail data is entirely contained within /var/spool/asterisk/voicemail. Are there any other files/directories I should be concerned with? Discussion around this topic in general appears to be somewhat spotty (and there's nothing on voip-info.org) so any comments or suggestions people have would be appreciated. -Steve
[Asterisk-Users] 1800 Provider
Hi list, I'm interested inreceiving incoming call to myAsterisk PBXthru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitationsuch asmuch few states that a call can originate? How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? TIA! Jim
RE: [Asterisk-Users] 1800 Provider
We can provide you 2.2 cents a minute 1800 number through SIP or H.323. We can also provide local access numbers and great worldwide termination rates. Regards, Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glendale, CA 91201 [EMAIL PROTECTED] tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jim OnnetSent: Saturday, May 08, 2004 1:11 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] 1800 Provider Hi list, I'm interested inreceiving incoming call to myAsterisk PBXthru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitationsuch asmuch few states that a call can originate? How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? TIA! Jim
Re: [Asterisk-Users] Transfering with Grandstream Phones
On Sat, 2004-05-08 at 20:43, Ryan Courtnage wrote: On 8-May-04, at 12:09 PM, Paul Tyreman wrote: I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have the same problem (attempting to transfer a call made to my BT102 will result in that call being disconnected/hung). Workaround is to use '#' to transfer instead of the 'transfer' button on the phone. I also agree.. Using the '#' key is the only way to transfer. I'm running Software Version: 1.0.4.63 Nothing in the html menu mentions how 'transfer' might work - perhaps its a blank key waiting to be programmed one day??? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
This hack is a tiny bit better: http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html John Baker John Todd wrote: At 10:31 AM -0400 5/8/04, Billy Huddleston wrote: That won't work.. That'll DIAL multiple phones/extensions, but will only bridge 1 of them when it auto-answers.. What we need is a way to have something like meetme call multiple extensions and bridge them to a meetme confrence (all of them muted but the admin of course, as it's a one way page) and then we would have a true paging system.. OK, I typically would badger people into looking in Google for this, but I'll be darned if I can't find this post on Google myself (search for Office-wide paging with Asterisk or AGI(callall) so I'll re-post here. This is a terrible hack. Someone _please_ make this cleaner. I'm looking at how to add this to the Wiki, but I don't see anything that's obviously marked as start new thread or similar links. If anyone is feeling ambitious, please add the stuff below. JT Date: Sun, 18 Jan 2004 17:22:11 -0700 To: asterisk-users-lists.digium.com From: John Todd [EMAIL PROTECTED] Subject: Office-wide paging with Asterisk and Cisco 7960 7940 phones I spoke the other day about my preliminary tests with office-wide paging with Cisco phones using the new SIP 6.1 image which supports auto-answer. I've got a small and crude recipe for those of you who want to experiment and hopefully create some better and more complete examples than the one I've thrown together below. Create a new line on each of the Cisco phones, and put the configuration into sip.conf as you normally would. I figure you have enough clue to create a new line in sip.conf and on your Cisco phones at this point. Go into settings - Call Preferences - Auto Answer (intercom) and then make the new line you've just created as auto-answer. (I wish there was a way to do this via the configuration file! Having to set this while sitting in front of the phone is silly and wasteful.) Now that you have created a valid Asterisk-capable SIP line that auto-answers, here's how you get the paging features to work: Here's what I have in extensions.conf: [conference] exten = ,1,AbsoluteTimeout(21) exten = ,2,AGI(callall) exten = ,3,MeetMe(,dq) exten = ,4,Hangup exten = t,1,Hangup exten = T,1,Hangup exten = h,1,Hangup ; [add-to-conference] exten = start,1,AbsoluteTimeout(20) exten = start,2,MeetMe(,dmq) exten = h,1,Hangup exten = t,1,Hangup exten = T,1,Hangup Here are the contents of /var/lib/asterisk/agi-bin/callall #!/bin/sh cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing Make sure to make the script executable. And then for every extension I have as an auto-answer, I have a file like this in /var/lib/asterisk/agi-bin : Channel: SIP/2006 Context: add-to-conference Extension: start Priority: 1 CallerID: Office Pager So, I have three lines that are configured for automatic answering - SIP/2006, SIP/2007, SIP/2008. I have three files named 2006-conf, 2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into the outgoing call spool directory every time I call extension . These three lines are the auto-answer lines on each of the three phone devices I'm experimenting with. Now, dial from any phone and you should have one-way paging. Voila! People who use the pager may have to get used to waiting 1-2 seconds before speaking to allow all the phones to catch up with the audio stream. All of the phones hang up after 20 seconds, regardless of if the person originating the page has stopped talking. Change the AbsoluteTimeout values to increase this interval. If you want a really confusing loud mess, then change the dmq options to dq and you'll get an N-way conversation going with everyone who has a phone. Bad. If you want a really interesting office surveillance tool, change the dmq to dt and you'll suddenly be listening to all of the extensions in the office, like some kind of mega-snoop tool. Useful for after-hours listening throughout the entire office. Someone should improve my scripts with the following changes: 1) AGI should automatically show the caller ID of the person originating the call instead of a generic pager address 2) The AGI should take arguments of what extensions to call and then dynamically create the list of files that get copied out to the /var/spool/asterisk/outgoing directory JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 - Gatekeeper - asterisk - SIP config problems
After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP - then I think this might work or must I put gnuGK on a separate machine? ps Documentation on the combination of Asterisk, h323 and Gatekeepers is really well hidden - I ain't seen it anywhere. In oh323.conf - I have the section... [register] context=h323phone alias=Call from gwprefix=0 gwprefix=1 gwprefix=2 gwprefix=3 gwprefix=4 gwprefix=5 gwprefix=6 gwprefix=7 1 - [h323phone] in extensions.conf is identical to my [sip] section (for my internal phones) - seems to work OK. 2 - the 'Call from' appears now with the CLID on the displays of the H323 handsets - can't I get it to show the users name of the Extension? 3- the gwprefix lists then seems to make asterisk the default gateway for the numbers dialled that start with [0-7] - so asterisk completes the H323 handsets call - this seems ugly - and I have not seen anyone else's config doing anything similar. What dumb thing am I doing? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Re: 729 licence on scsi
Not with the voiceage system, but with the new system you will be able to. Mark On Sat, 8 May 2004, nicolas wrote: if i have ordered one lic. and now i have realized i need two lic for one call (2 cannels one to provider one to sipphone) can i install 2. lic with another reg code ? nico Mark Elkins wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? Who do I talk to now? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
Don't know how far you've tried to take the 1204 in terms of functions, but we did the same thing over a two month period and found: 1. handling outbound calls on a per pstn line basis (eg, directing certain calls to certain pstn lines) is very non-standard and subject to future failures as code changes happen in * and the 1204. Correct, but I don't need to have access on per channel basis. 2. ring-cadence detect is done on the first ring after the 1204 reboot and applied to all four ports. If the pstn lines happen to come from different Central Offices (with slightly different cadences), callerid and other such timing sensitive functions will fail. I believe that you can actually change that, you have to specify time in ms not a number of rings. 3. security is less then acceptable. If the 1204 is exposed to the Internet, anyone can make calls, change settings, etc. Correct, but in real production wouldn't you keep it behind the Firewall? 4. the per-port cost is substantially higher then many other products if you consider the cost of keeping the firmware reasonably current as standards evolve. Yes, but SIP connectivity (instead of PCI) adds lots of flexibility 5. the box does not follow published sip standards; only selected pieces. I am sure that they will release new firmware with better support for SIP 6. diagnosing problems and monitoring operational functions in a real-world production environment is less then acceptable. Agreed 7. support is limited to whatever your reseller provides, which is less then acceptable if your reseller is not familiar with *. This is reality of the 21st century :) We also found the voice quality to be very good, echo cancellation was good, etc. With relatively easy firmware tweeks to interoperate with * and standards better, it would be a nice pstn interface; however, they seem to not have any interest in going there. :) Regards, Wojtek Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 (4x FXO)
Their current firmware doesn't allow to write to the section for SIP registration. I am able to communicate with itbydialing [EMAIL PROTECTED]. Also, you have to protect this box with Firewall otherwise the whole world will be able to call through it. Regards, Wojtek - Original Message - From: Dawid Mielnik To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 9:40 AM Subject: RE: [Asterisk-Users] Mediatrix 1204 (4x FXO) And what problem do you have with registering ? Jeremy Jones has recently posted his SNMP walkthrough from a mediatrix 1104 - you might reference that, configuring 1204 should be very similar to that of 1104. Regards, Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Wojciech TrycSent: Thursday, May 06, 2004 5:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mediatrix 1204 (4x FXO) I have successfully implemented 1204 in semi production environment. Just want to share that it works very well, through the firewall (NATed). Unfortunately, it can not register with the server (and authenticate) but otherwise everything is fine. The audio quality is very good. Regards, Wojtek
[Asterisk-Users] DIAL without connect...
Hi, I want to dial a sip device from extensions.confbut not connect the other party when it picks up, so I can send some DTMF before I connect the call. Is that possible ? Thanks a lot! HQ
Re: [Asterisk-Users] DIAL without connect...
On Sat, 2004-05-08 at 16:28, HCQ wrote: I want to dial a sip device from extensions.conf but not connect the other party when it picks up, so I can send some DTMF before I connect the call. Is that possible ? You must have missed this message: From: Eric Wieling [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.64,1.65 Date: Fri, 07 May 2004 15:34:04 -0500 On Fri, 2004-05-07 at 16:30, [EMAIL PROTECTED] wrote: Update of /usr/cvsroot/asterisk/apps In directory mongoose.digium.com:/tmp/cvs-serv17955/apps added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call + 'D([digits])' -- Send DTMF digit string *after* called party has answered\n +but before the bridge. (w=500ms sec pause)\n I nominate anthm for Asterisk Sainthood. This is a feature that people have been asking about for a LONG time. Thanks anthm! -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100p / Answer- Flash - Dial
I have an X100P connected to an extension of aPanasonic PBX.When a call from the PSTN comes in,it is routed directly to theextension where the x100p is .I want* to answer the call, play amessage and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then todialan IAX channel. The problem is that when I do a Flash on thezap channel, and then try to dial a new extensionvia that zap channel I get the following error "can't createzap channel". If I do a SendDTMF()thecalldoes get transfer to the new extension but then * gets out of the callloop and don't know it is answered or not by the new extension. AmI missing something? Why am I getting the "can't creatza channel" Thanksin advance. Dan
RE: [Asterisk-Users] x100p / Answer- Flash - Dial
Title: Message Even if you could get that to work properly, which I dont know... the callprogress detection is horrible; if you want to do that reliably you need a T1,ISDN or IPinterface to the switch (something that actually provides proper call progress) Sam -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan FernandezSent: Saturday, May 08, 2004 11:44 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] x100p / Answer- Flash - Dial I have an X100P connected to an extension of aPanasonic PBX.When a call from the PSTN comes in,it is routed directly to theextension where the x100p is .I want* to answer the call, play amessage and then transfer the call to another extension via the Zap channel where the call was received (I need to flash the zap channel) . If this extension doesn't answer I want then todialan IAX channel. The problem is that when I do a Flash on thezap channel, and then try to dial a new extensionvia that zap channel I get the following error "can't createzap channel". If I do a SendDTMF()thecalldoes get transfer to the new extension but then * gets out of the callloop and don't know it is answered or not by the new extension. AmI missing something? Why am I getting the "can't creatza channel" Thanksin advance. Dan smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] asterisk with german SIPGATE ?
hi anybody running with german SIPGATE? my configuration don't works :-( regards [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stripping numbers at the end of a dial pattern = extensions.conf
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
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[Asterisk-Users] Low Bit Rate Codecs
Greetings all, I have searched all over and have found bits and pieces on low bit rate codecs, however I have found it very difficult to compare apples with apples. The conclusions I have come to are as follows, I would appreciate if anyone has some feedback, or point me to where I might find this sort of comparison in black and white G723.1 very low bit rate used commercially, not avail for * (I am currently using this codec in another commercial application and therefore it is my reference point) G729a low bit rate slightly higher bandwidth usage than 723.1 ??? avail as a low cost add-on for * better quality that g723.1 ??? iLBC Low bit rate slightly higher bandwidth usage than 723.1 and 729a ??? open source, no additional cost for * quality comparable to G729a stands up better ip paths suffering from latency and jitter ??? any comments ?, or any codecs I have missed in the same class ? craig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with german SIPGATE ?
was posted on a day or two ago Thorsten Gehrig wrote: hi anybody running with german SIPGATE? my configuration don't works :-( regards [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1800 Provider
Jim Onnet wrote: How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? By lying to their customers about the actual rate they are being charged. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAL without connect...
Asterisk CVS instructions: http://www.asterisk.org/index.php?menu=download Download CVS HEAD or CVS STABLE for Asterisk. Zaptel and LIBPRI do not have a CVS stable branch, only a CVS head branch. I recommend the stable branch. The code you are looking for is only in the HEAD branch. On Sat, 2004-05-08 at 17:44, HCQ wrote: CAn you help me on how to take that code out? I tried with CVS export but it says there is no directory with that name.. Tx. HQ. - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 08, 2004 6:40 PM Subject: Re: [Asterisk-Users] DIAL without connect... On Sat, 2004-05-08 at 16:28, HCQ wrote: I want to dial a sip device from extensions.conf but not connect the other party when it picks up, so I can send some DTMF before I connect the call. Is that possible ? You must have missed this message: From: Eric Wieling [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-cvs] asterisk/apps app_dial.c,1.64,1.65 Date: Fri, 07 May 2004 15:34:04 -0500 On Fri, 2004-05-07 at 16:30, [EMAIL PROTECTED] wrote: Update of /usr/cvsroot/asterisk/apps In directory mongoose.digium.com:/tmp/cvs-serv17955/apps added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call + 'D([digits])' -- Send DTMF digit string *after* called party has answered\n +but before the bridge. (w=500ms sec pause)\n I nominate anthm for Asterisk Sainthood. This is a feature that people have been asking about for a LONG time. Thanks anthm! -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1800 Provider
60 second increments, per call connect charges that range in the 39-99 cent range. bkw - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 08, 2004 4:57 PM Subject: Re: [Asterisk-Users] 1800 Provider Jim Onnet wrote: How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? By lying to their customers about the actual rate they are being charged. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1800 Provider
They also structure the fees in such a way that it is impossible to actually use the full value on the card. IIRC, they term this breakage and it means you end up with at least some amount of unusable value left on the card at end of use. THX/BDH On Sat, 2004-05-08 at 20:19, brian k. west wrote: 60 second increments, per call connect charges that range in the 39-99 cent range. bkw - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 08, 2004 4:57 PM Subject: Re: [Asterisk-Users] 1800 Provider Jim Onnet wrote: How do the phone cards company with 2cents/minute rate do it by giving out 1800 access number? By lying to their customers about the actual rate they are being charged. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P keeping PSTN line Offhook
Try enabling busy detect and set it to a value between 4 and 6. If you set it too low you might start getting random call drops. I think this problem is due to some providers allowing only the called party to hang up. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shahid Sent: Saturday, May 08, 2004 6:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P keeping PSTN line Offhook Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! pbx1*CLI zap show channel 1 Channel: 1 File Descriptor: 31 Span: 1 Extension: Context: bell Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Offhook = zapata.conf == busydetect=no musiconhold=default group=1 pickupgroup=1 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel = 1 pickupgroup=1 immediate=no signalling=fxs_ks callerid=asreceived channel = 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Bit Rate Codecs
Craig wrote: Greetings all, I have searched all over and have found bits and pieces on low bit rate codecs, however I have found it very difficult to compare apples with apples. The conclusions I have come to are as follows, I would appreciate if anyone has some feedback, or point me to where I might find this sort of comparison in black and white G723.1 very low bit rate used commercially, not avail for * (I am currently using this codec in another commercial application and therefore it is my reference point) G.723.1 is pretty much obsolete. You don't see it being used on anything new. Most people do VoIP using RTP. The overhead of RTP is so huge, a small saving on the codec makes little difference. People generally go for G,729 now, which sounds considerably better. If you compare the total bit rate for G.723.1 vs G.729 in RTP G.723.1 often comes out considerably lower. This is because it works in 30ms blocks - you only have 33 RTP packet overheads per second. You can choose to pack more G.729 data into each RTP packet and even this up. The patent licencing for G.723.1 is a PITA, which hasn't helped it achieve widespread use. There are two variants of G.723.1, with different bit rates. The lower bite rate (5.something kbps) sounds nasty. The higher rate (6.something kbps) sounds more reasonable. Using 30ms blocks, it is not so compatible with *, which is geared to 20ms block processing. A lost packet causes a 30ms hole, so it tends to be less tolerant of packet loss than something working in smaller blocks. It sounds awful for anything but a single pure voice. G729a low bit rate slightly higher bandwidth usage than 723.1 ??? avail as a low cost add-on for * better quality that g723.1 ??? Definitely better quality than G.723.1. This is definitely the mainstream right now for VoIP. It is heavily patented, so free codecs are not possible. There are several bit rate options, but almost everyone uses the 8kbps variant. This sounds pretty good for its bit rate, though I think there have been better codecs. In telephony you need use something compatible with the far end, and G.729 seems to be the current common ground. It is rather intolerant of packet loss. Some people pack several G.729 blocks into a single RTP packet, to decrease the RTP overhead. That makes it even less tolerant of packet loss. It sounds awful for anything but a single pure voice. iLBC Low bit rate slightly higher bandwidth usage than 723.1 and 729a ??? open source, no additional cost for * quality comparable to G729a stands up better ip paths suffering from latency and jitter ??? iLBC has a much higher bit rate than G.729, but the voice quality is about the same. Why does that make it interesting? Well, it is designed to be much more tolerant of packet loss, and that makes it take more bandwidth. The design of RTP makes that take so much overhead that the total bit rate using iLBC isn't a huge jump from using G.729. However, if you use a more efficient streaming mechanism - say IAX, or an RTP like format with many calls packed in a packet - the total bit rate difference starts to look wider. There, the increase in bits is so great its quite likely to be the *cause* of packet loss, by clogging up the channel. :-) Good old GSM 06.10 is worthy of consideration. Free of patents (at least ones being actively pursued). Low compute requirements. Reasonable voice quality. Somewhat more tolerant of background noise than the codecs above. Although GSM networks don't use it much these days (they mostly use the newer EFR and half rate codecs) it's still a very servicable codec. Its bit rate lies between G.729 and iLBC. On a pure voice it gives poorer quality than G.729. Add some background noise and it can beat G.729. Its tolerance of packet loss is probably similar to G.729. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 500ms usleep in rtp.c ?
http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(50); to usleep(5000); in rtp.c the proble totally goes away...even the note above it says it needs to be fixed. bkw
