Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents
Why is it you have to put down the United States? Sprint is a U.S. Company. Vonage a U.S. Company. Digium a U.S. Company. What in the heck does it have to do with you? Even if Vonage is tied up in court the rest of the world doesn't care. VOIP will live on. We are not sue happy, This is big business 101. Sprint wants Vonage. This is a way to own them, grab their market, and muck up the rest of the telcos.. There is more here than meets the eye. We'll all have to wait and see what happens. But don't put us down, There are many great people living and working here including the creators of Asterisk. Matt Riddell wrote: trixter http://www.0xdecafbad.com wrote: Anyone thinking about doing a VoIP business may want to get more info before proceeding since they may not have the millinos vonage has to fight this. Unless of course they don't live in the United Sue'ers of America. :D ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on windows
Good explanation Rich. Unix was built for the riggers of the Telecomm industry. You won't find Windows running the PSTN. Unix and Linux are used where their needed for real time processing and the highest reliably. Windows is a productively OS that is easy to use for non technical people. I use both as do many of us. Each has there purpose. Rich Adamson wrote: Any of the more current Win32 systems can be programmed to handle near real-time events (eg, sip, rtp) just like linux, bsd, and other O/S's. Obviously, Call Manager is one such system. It's really not an O/S religious war/discussion, but rather a lack of knowledge (on any O/S that a poster might not be familiar with) on how to design/implement it in code. With that said, porting the low level drivers (zaptel, wctdm, etc) from linux to Win32 is no where near a trevial task, and would basically involve a complete rewrite of such code. Since there are very few people (maybe one or two) that truly understand _all_ the interworkings of the linux-zaptel drivers, and, I venture to guess those same people are not even remotely cognizant (no offense intended at all) of how to write Win32 drivers, don't look for asterisk to be fully ported to the Win32 environment any time soon. As far as I'm concerned, there isn't any real justification to do so either. A pbx is intended to be a near real-time system and as such should not have programmers/technicians mucking with it in a production environment. That also suggests that any form of GUI interface that is resident in pbx s/w is not only not required, but not desirable as it will lead to someone mucking with it and impacting availability. Running a GUI interface via a manager (cti or whatever) interface that is not part of the real-time pbx environment certainly is doable and has been done on lots of pbx and central office switches over the years regardless of what the underlying O/S happens to be on the switch. Those companies that have implemented near real-time systems have probably questioned their choice of O/S years after deploying production systems, but that's perfect 20-20 hindsight. Cisco (as only one example) tends to purchase the majority of their non-core products from other companies (or purchase the entire company), and in a fair number of cases, will attempt to enhance/port that product to something different generating significantly more negatives then if they would have left the product alone. I'd be one that would certainly stay away from the port of CCM on another O/S for at least a year. Rich been following this for a while, just thought I would add a bit to the debate, but doesn't the Cisco system (Call Manager?) run on an Windows 2000 based server - if it was that bad why would Cisco choose to run it? Also 3Com use NT/2000 to run the H323 gateway. Admittedly the call processor runs on VXWorks but to cross the boundary of proprietary 3com and rest of world - they jump onto windows. Curiously Wayne. ps I don't know a great deal about the cisco system - its more hearsay so please jump in on :) Patrick wrote: Reminds me of an Internet Call Diversion pilot WorldCom did back in 2000 where Alcatel some M$ drones brought in 2 very big Alpha servers running NT. These boxes needed to be rebooted multiple times. They were surprised WCOM felt having to reboot these boxes all the time was unacceptable in an environment requiring 5nines availability. Never laughed so hard when I saw the incredulous faces of the M$ drones. We brought in a Stratus based solution and won the project. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and RTP streams
Sherwood, I have never known the RTP audio to be on only one port in sip. I believe it's always on 2. The one way audio is always a nat/firewall problem in sip. Sherwood McGowan wrote: Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] change codec based on callerid (sip/iax)
This can be done by modifying the source code. trixter http://www.0xdecafbad.com wrote: I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
Don, I agree with you on many fronts. I come from a radio background and here in southern cal unless we fall into the sea nothing will take out all of the communications here including ham because we are not in low lying flat land and were too diversified, over 150 miles and as many mountain top sites. BUT, let me tell you about how bad the southern CA. radio site owners are becoming. We had a 4 day outage at a very large site where one of my radios is located. None of them care anymore about backup power. This happened this past week. We took up our own Generator because the site owner (a national site company) won't maintain an old one. My friend (a microwave isp ) fixed the site owners by adding oil and a new battery. That will take us out! Don Fanning wrote: Time and time again, emergency action drills take place in cities to target where their weaknesses are in "crisis" handling. Usually they involve planes crashing or explosions (mock of course). Obviously they were never prepared for this sort of disaster in their recovery plan. I've participated in a few ARES/RACES drills and have to say that much could be done to improve upon the "HAM" infrastructure. Most of the time, communications is coordinated through 1 repeater system. When this repeater goes down, of course people would switch comms to another but in a case like this, where all the repeater systems go down except for maybe one, there needs to be a better plan. In Amateur Satellite Service, these orbiting "Repeaters" employ a system called RUDAK where a chunk of spectrum is repeated. Obviously this isn't feasible in terrestrial repeaters but they dohave the ability to turn off radios and switch bands at will depending on operating conditions. With software controlled radio and Asterisk, the repeater system could be made to be more resilient to disaster by linking to other repeater systems via radio where it could connect outward. If you figure the overhead of a repeater's transmitter and receiver plus the controller, replaceing the controller with an asterisk based unit (integration) would make more sense as it would give the repeater system much more capabilities in the same footprint and power. Additionally, these repeater systems are located on hilltops with other radio systems so they should have emergency power available (if you've ever been to a hilltop repeater site, you'll know what I mean). I think the biggest thing that hurts ham radio's ability to react to a crisis is the lack of equipment and operators. Most of the traffic we pass is "Health and Welfare" with "Logistics" being the second to it. What defeats this is that in a disaster where local/high band long haul capabilities are diminished, is simply the one repeater that is functional because everything is squeezed onto one VHF/UHF repeater. Where I could see thing being improved? Installation of 802.11b/g WLAN under Part 97. It would allow for more users into the system, there are less hardware and power components and allows the system to be dynamically configured. Asterisk could play a huge role then as it's made for IP based traffic and could re-route in a split second. -Don From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael D Schelin Sent: Saturday, September 10, 2005 10:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM The two best forms of communications in a real disaster and one always has been is #1 Ham radio. and #2 satellite telephone. Ham radio is global and has proven time and time again to be the most reliable when the infrastructer has been damaged. The U.S government is the biggest user of satellite telephones which is also becoming a valuable tool again when the communications infrastructure is down. It would be nice If Asterisk could be used but in this case but it's useless. People are displaced and most of the communications infrastructure for the city is unusable. I don't mean all of the telco's systems. It's the flood that wiped out most home and business systems. For us, The best thing that a provider can do is to have redundant servers in different cities. This should remind us all how fragile our lives are. Chris Travers wrote: Mark Phillips wrote: Hold on here folks, I'm guessing that the original poster of this thread isn't a member of his local RAyNet team. Whilst I don't profess to be an expert at this I have been doing emergency radio for quite some time and have seen service at the Lockerbie bombing, Docklands bomb, Ground Zero (I'm sure I'm a terrorist target y'know - they seem to follow me everywhere) and soon I'll be in Louisiana. In all of these events the KISS principle must and does prevail. We need a system that is a simple and energy efficient as possible.
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
The two best forms of communications in a real disaster and one always has been is #1 Ham radio. and #2 satellite telephone. Ham radio is global and has proven time and time again to be the most reliable when the infrastructer has been damaged. The U.S government is the biggest user of satellite telephones which is also becoming a valuable tool again when the communications infrastructure is down. It would be nice If Asterisk could be used but in this case but it's useless. People are displaced and most of the communications infrastructure for the city is unusable. I don't mean all of the telco's systems. It's the flood that wiped out most home and business systems. For us, The best thing that a provider can do is to have redundant servers in different cities. This should remind us all how fragile our lives are. Chris Travers wrote: Mark Phillips wrote: Hold on here folks, I'm guessing that the original poster of this thread isn't a member of his local RAyNet team. Whilst I don't profess to be an expert at this I have been doing emergency radio for quite some time and have seen service at the Lockerbie bombing, Docklands bomb, Ground Zero (I'm sure I'm a terrorist target y'know - they seem to follow me everywhere) and soon I'll be in Louisiana. In all of these events the KISS principle must and does prevail. We need a system that is a simple and energy efficient as possible. Building a network of * servers and Wi-Fi links is all very well but how are you going to power them? These are excellent points. I have a few interesting suggestions here The first is that the only obstacle to any sort of longer-range point to point line is merely power. This is true whether you are talking HAM or fiberoptics. Note that if you have the power, it would take disruption of the physical line to disrupt a fiber line. Note that DirectNIC in New Orleans remained operational without *any* downtime or loss of connectivity with the rest of the world. The suggestion that I have is for various areas to have dedicated civil emergency com units with strategic reserves of fuel (3-4 weeks worth), battery backups, etc. These units would have links (fiber, microwave, and/or satellite, better to pick 2 of 3) to areas outside expected disaster zones. Asterisk could then run across these links. (Sattelite links would best be POTS-type). The point is to a disaster-tolerant communications infrastructure which could then be used to to provide additional communications services to the relief workers. With various point to point wireless capabilities, it might be possible to use them to provide cell service to relief workers etc through the installation of GSM microcells (which could be brought in after the fact). See where I am going? Generators require fuel which is always in short supply and batteries die out quickly. Adding Ham Radio to the picture doesn't really add much when you are trying to do something like a * network. The radio gear just isn't designed to integrate with the * server. Ham radio is being used down in the Katrina affected area with great results for both emergency and heath/welfare related traffic. They are using both "phone" (that's when one talks in to the radio) and data modes and can be heard all over the 75 and 40 meter bands here in the US. Power for most of these stations comes from batteries they loot (with Police approval) from abandoned cars or a combo of solar and batteries. Many stations are only hear on the air after dark so that they can put as much sunlight into their batteries as possible. Yes, electricity is available in some places either all day or across the peak hours (allowing the workmen to restore power to other areas). Yes, there are radio to phone interconnects but these really are a single phone to a single radio. Think of it as a cordless phone in that the radio user can be anywhere within reach of the base station. Such technologies, whilst legal here in the US, may not be legal elsewhere. When last at home (UK) I was not able to connect my radio to the phone system by law (this may have changed recently - not been home for 8 years). Many countries have such restrictions and as we saw during the Tsunami, rules don't get relaxed just because there's a panic on. Without question a phone system would be much better than a radio station. As such I'll be taking a portable * server I've built, all the IP hard phones I can find and 5 DirectTV style Internet systems. How do IP hardphones work with satellite internet? I always thought people had real trouble getting them to work at all. Best Wishes, Chris Travers Metatron Tecnology Consulting ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk
Ben, That is the correct choice for an Asterisk box. good luck. Ben Brown wrote: Thanks for the replys. I'm convinced. PRI it is. Peter Svensson wrote: On Mon, 5 Sep 2005, Ben Brown wrote: So the only difference with PRI is caller ID? What I am trying to determine is if the PRI has enough advantages to give up the voice channel used by the D channel. For what I am doing, caller ID is not necessarily that important for my application. The PRI signalling is more robust than any of the alternatives (except SS7). Call setup is faster, you can get DID, caller id and much better error reporting from the pstn. I would recommend against CAS or analoge connectes whenever isdn is available. Can Asterisk choose the context based upon the CallerID with a PRI? Yes, this can be acclomplished in the dialplan. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk
Go T1 with PRI signaling. Farming and line coding is for all T1's. We use ESF (extended super frame) B8ZS ( I forget B8ZS stands for but it's a newer line coding) . If you have it avaible to you, Signaling type should be PRI. The rest of your numbers 4-7 are in the PRI signaling. No sound differences in digital. Caller ID is very important. PRI signaling is very easy to set up with Asterisk. Ben Brown wrote: Preparing to order a T1 (not PRI) for our asterisk box. The telco has offered me several options that I am not sure of. Which would be best for use with asterisk? The box has the Digium card in it, BTW. 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, EM, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start - Wink, Immediate, Seizure 7. If Seizure then - Origination, Digit Collection. On a related note, am I correct that the only major differences with a PRI are faster call setup time and the caller ID information on the D channel? Are there any significant differences in sound quality with a PRI? Any other advantages to giving up the extra channel seeing as the cller ID is not really a selling point for me? Thanks BEN ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channalized T1 and PRI with Asterisk
Is a non-PRI T1 significantly harder to configure with Asterisk? I don't think so. I moved to SS7 signaling and convert down to other formats. I've used CAS signaling for years and it has worked just fine. But not with Asterisk. Remember That CAS signaling is an inband format and requires more processing to set up a call. PRI signaling (out of band) is used now more often because it's easy to set up not just in Asterisk but at the CO. There is less for to go wrong for a Telco. Most of them have moved away from the older signaling formats for that reason. If you just can't live with 23 channels and you need the 24th. and your Telco is willing to give you CAS signaling then go with it. My old set up is EM, DTMF,Wink,Origination 7 digits. Some telcos will not out pulse in DTMF. I don't know if Asterisk will accept MF. Like I said If you can get PRI then get it so you don't have this mess. Ben Brown wrote: So the only difference with PRI is caller ID? What I am trying to determine is if the PRI has enough advantages to give up the voice channel used by the D channel. For what I am doing, caller ID is not necessarily that important for my application. Is a non-PRI T1 significantly harder to configure with Asterisk? Can Asterisk choose the context based upon the CallerID with a PRI? Thanks for your reply BEN Michael D Schelin wrote: Go T1 with PRI signaling. Farming and line coding is for all T1's. We use ESF (extended super frame) B8ZS ( I forget B8ZS stands for but it's a newer line coding) . If you have it avaible to you, Signaling type should be PRI. The rest of your numbers 4-7 are in the PRI signaling. No sound differences in digital. Caller ID is very important. PRI signaling is very easy to set up with Asterisk. Ben Brown wrote: Preparing to order a T1 (not PRI) for our asterisk box. The telco has offered me several options that I am not sure of. Which would be best for use with asterisk? The box has the Digium card in it, BTW. 1. Dial Tone - No, Yes - Precise, Yes - SCC 2. Framing - SF, ESF 3. Line Coding - AMI, B8ZS 4. Signaling Type - Ground Start, EM, Loop Start w/Ring, Loop Start w/o Ring 5. Pulse Mode - DTMF, MF 6. Outpulse Start - Wink, Immediate, Seizure 7. If Seizure then - Origination, Digit Collection. On a related note, am I correct that the only major differences with a PRI are faster call setup time and the caller ID information on the D channel? Are there any significant differences in sound quality with a PRI? Any other advantages to giving up the extra channel seeing as the cller ID is not really a selling point for me? Thanks BEN ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
The Asterisk Software is not the problem. I'm thinking and I could be wrong that your having a total line balance mismatch with the card your using. Check the line impedance and the card's. Most people using Asterisk don't have that much echo. Anyway It would be nice to see a manual Hybrid adjustment on analog cards. Don't give up. canuck15 wrote: I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production parts. I bought a couple different SIP phones to try and a Digium TDM01B card. I am using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MBfor my test system. The only thing that will be different on a production system is that I will be using a newer chipset PC with faster processor and 512MB. Probably Intel 7505, 7210, or 7211chipsets which seem to be the most compatible with Asterisk. My problem is that I cannot eliminate echo no matter what I try. I seriously doubt that a newer chipset faster PC with more memory will eliminate or even reduce my echo problems based on what I have read.I am not about to drop more cash to try and find out. Essentially, my findingsare that Asterisk is NOT production capable for my configuration which is via FXO and PSTN. That is probably THE most common configuration so if itis not production capable like that itisn't production capable period as far as I'm concerned. What a disappointment :(. Unless I am missing something I am sure that many many people with a similar configuration in a production environment have the same problem. Perhaps they are just living with it?? For me it is just as unacceptable on an Asterisk system as it is on a traditional PBX. Some calls are ok and some are not. No correlation to local, long distance, time of day. There always seems to be some echo. Sometimes it is worse than other times. Again, no correlation to local, long distance, time of day. Tried connecting to ATA adapter and using VoIP provider instead to see if the telco was causing the problem. That did not change anything. Still the same general echo problem The things I have tried includein no particular order and not limited to are: *Buy latest TDM400P withlatest FXO module *Ensurecopper connection to analog telco lines and telco are not causing problems including running a separate shielded line to the demarc AND having the telco guy come out and test the levels, impedance etc. *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method and by using the detailed Ztmonitor method via a Telco 102milliwatt test phone #. The end result was RX=8.0, TX=-1.0. Since I still have echo problems I have tried all sort of other settings without success. *After ALL of the above, try every possible combination of all of the following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 2(default, aggressive,CVS head developments, bugs.digium.com patches, adjust threshold level as per wiki etc. etc.) *Make sure echotraining line is before FXO channel assignment in zapata.conf file *Run fxotune which did not find a need to adjust the FXO levels (1=0,0,0,0,0,0,0,0) Based on all the above testing the best settings were pretty much in line with what most people are finding. echocancel=on. echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, TX=-1.0. Still have echo. Aggressive mode helps a bit but then the other persons voice get's cut offa lotespecially when I talkand the cutting in and out of the canceller is more noticeable and objectionableingeneral thanif Aggressive is turned off. Ihave two SIP phones. An Aastra 9133i and a Grandstream GXP2000. Echo problem is the same on both phones. I am located within a metropolitan area in Canada. Any comments and/or suggestions would be greatly appreciated as I am pretty much out of ideas and ready to give up on Asterisk as a suitable traditional small business phone system replacement. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Register Today for Fall 2005 VON: The Destination for IP Communications
I don't think this will work but it's worth a try. Fall VON 2005 http://von.com is happening September 19-22, at the BCEC in Boston. As usual, we have a special offer for members of the pulvermedia community, which is valid for the month of June only. Register using priority code JUNE and save an additional $300 off current pricing for either the VON Package or Full Conference registration options. In addition, you can register for FREE access to the exhibit hall using the JUNE priority code. To register, please visit: http://von.com/register.html Dean Collins wrote: Anyone able to get me a comp/highly discounted ticket to this? $150 just to visit the exhibition halls sounds crazy? Dean -Original Message- From: Jeff Pulver [mailto:[EMAIL PROTECTED]] Sent: Tuesday, 23 August 2005 11:47 AM To: mailinglist1 Subject: Register Today for Fall 2005 VON: "The Destination for IP Communications" Hi There, While flying to London yesterday, I spent some time thinking about VON and how while some things change, other things about VON remain the same. Since our first VON event in the Spring of 1997, our VON events have over time become the worldwide Destination event for IP Communications. In fact, while we are actively marketing Fall 2005 VON using various channels around the United States, it is the continued strong word-of- mouth buzz that is bringing in delegates from around the world. So far, there are delegates registered from 40+ countries including: Argentina, Aruba, Australia, Austria, Belgium, Brazil, Canada, Chile, China, Costa Rica, Denmark, Dominican Republic, Finland, France, Germany, Ghana, Hong Kong, Hungary, India, Ireland, Israel, Italy, Japan, Korea, Mexico, Netherland Antilles, Netherlands, New Zealand, Norway, Russia, Singapore, Slovenia, South Africa, Spain, Sweden, Switzerland, Taiwan, Turkey, UK, UAE, USA and Uzbekistan. I expect the buzz to be pretty strong when the doors open in less than four weeks. The 330+ exhibitors in our "Sold Out" exhibit hall represent our largest exhibit hall...ever! (and has grown by more than 100 exhibitors since Spring 2005 VON.) The Fall 2005 VON conference sessions are returning to the size we experienced five and six years ago. The registered delegates in Boston are all part of the ecosystem that makes up our VON events. There will be people representing just about all aspects of the IP Communications food chain. Note: Vendors who are interested in exhibiting at Spring 2006 VON should consider signing up now. The pulvermedia Sales team is projecting that the exhibit hall at Spring 2006 VON will be close to sold-out before we arrive in Boston for the commencement of Fall 2005 VON. Experience the Journey and register today for Fall 2005 VON, "The Destination for IP Communications." Please visit: https://secure.pulver.com/von/register.html to register. Best regards, Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't get G729 working after buying a license.
Call Digum. They support the license codec install. Matthew Schumacher wrote: List, I purchased 2 g729 licenses but I can't get it to answer a g729 call from a cisco router with a vwic card. In the debug output below you will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) when it should support g729 according to the config also listed below. The real odd thing is I can place g729 calls to the router, just not from the router to *. Anyone have any ideas on how to fix this? Another problem I am having is I want to use the info dtmf mode, but the sip packet that asterisk sends does not announce info in the Allow string. Thanks, schu in debug: 20 headers, 13 lines Using latest request as basis request Sending to 192.168.77.254 : 5060 (non-NAT) Found no matching peer or user for '192.168.77.254:49206' Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 19 Peer audio RTP is at port 192.168.77.254:16494 Found description format G729 Found description format telephone-event Found description format CN Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (g723), peer - 0x3 (g723|gsm), combined - 0x1 (g723) Aug 23 09:54:43 NOTICE[1379]: chan_sip.c:2792 process_sdp: No compatible codecs! Transmitting (no NAT): SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.77.254:5060 From: sip:[EMAIL PROTECTED];tag=4194CB3C-F91 To: sip:[EMAIL PROTECTED];tag=as4ebd30b1 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 in sip.conf: [router] type=friend context=default host=192.168.77.254 dtmfmode=info disallow=all allow=g729 nat=no canreinvite=yes qualify=yes in debug: [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx == Found license 'G729-' providing 2 channels == Found total of 2 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 2 == Registered translator 'lintog729' from format slin to g729, cost 11 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overriding Caller ID
PRI is one of many signaling formats available on a T1 circuit. T1 and
PRI are not the same thing. Have your carrier change the signaling
format of your T1 to PRI. PRI is 23 B Channels and 1 D Channel
(Signaling) and is an end office protocol. It has many of the features
of a full blown SS7 network.
Waldo Rubinstein wrote:
Derek/Jeremy/Kevin,
Thank you all for your comments. I suspected the issue would be the
fact that we don't have a PRI but a T1. However, I decided to post
the question to the list simply because I would assume that because
the carrier is looking at our circuit configuration when answering my
questions, they should know we don't have a PRI and they should have
told us that we need PRI or it won't work.
To address your previous post, we are setting it to a caller id of a
number assigned to us.
I'm just confused simply because of the fact that the carrier never
caught or mentioned that we don't have a PRI and I spoke with 4 of
their engineers and 2 supervisors. Go figure. I guess it's time to
switch to a better carrier that knows what the heck they're doing.
Thanks again,
Waldo
On Aug 19, 2005, at 5:35 PM, dbruce wrote:
Hmm... I missed what Kevin and Jeremy caught...
But... you did mention TE410P and T1 circuit from the provider...
So, what exactly is between the TE410P and the T1 circuit... As you
have it
configured, it should not work at all
If the far end switch is a DMS100 you should have signalling=pri_cpe.
Regards,
Derek
- Original Message -
From: Waldo Rubinstein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 19, 2005 2:51 PM
Subject: [Asterisk-Users] Overriding Caller ID
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible combinations. When speaking with FDN,
they say they have set their T1 to show our main number for outbound
calls, but that we should be able to override that with no problem.
As I said, I have tried all possible combinations, yet, nothing seems
to work. Below are snippets of some of our configs:
extensions.conf
;
; Local calls
;
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCallerID(2125551234)
exten = _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel = 1-24
Does anyone have any suggestions?
Thanks,
Waldo
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Re: [Asterisk-Users] Full T38 sip Faxing now Available
The Multitech mvp24xx and the 130 are true T38 devices and work well. Cory Andrews wrote: We use the MultiTech FaxFinder 100 and 110 (1 and 2 port fax, respectively). We have it integrated with an Asterisk server, faxes are routed to the FaxFinder, converted to PDF files and sent to an electronic depository. Documents are faxed electronically from the desktop. Not really a solution for a large enterprise, but we have a 2 port unit that services our inside sales department and everyone seems to like it. Cory Andrews VOIPSupply.com v – 716.630.1555 X22 e – [EMAIL PROTECTED] Chris Mason (Lists) wrote: If you have a good NAT device (like a PIX) then it should. We got 1 to go thru by doing NAT=no and canreinvite=yes on an ATA that was behind a PIX. We are trying again now to force a static public IP to an ATA and see if it works that way. Right now we can't get the re-invites to happen. I'll take that as a no then, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Full T38 sip Faxing now Available
I'm not hiding anything from this user group. Buy a Cisco gateway and put in you own T38 network together. I can't respond that fast to the hundreds of emails I've receive. All I'm saying is if you want T38 now then buy our service. If not, then wait for the Asterisk community to release it. If you want to speed up the process then help develop the code. T38 is not for a beginning programmer. I posted this to help the people who are under a gun to get something working now. It will be great when the real Asterisk developers release it. We're waiting just like you but in the mean time don't flame me for informing the group about something that can help them. Marc Storck wrote: Do you want to share your knowledge how to get it work??? Regards, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Full T38 sip Faxing now Available
Why do you put me down? I have not done a thing to you and I'm not a spammer. Please stop this activity It's not professional. If I were to give you bad service please feel free to comment negatively but I've never dealt with you nor do you have an account with us. Sincerely Michael D. Schelin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: two UA with the same usr/pwd
Rich is correct. Example: Night security guards may need to catch an inbound calls that could ring at more than one station. Maybe one is doing rounds and the other is at another desk off site. Sometimes call forwarding is too slow. There are many reasons why this could be used. Rich Adamson wrote: Regardless of what has (or has not) been implemented in asterisk, there is a very valid business reason for wanting an extension number to ring on multiple phones and to determine the status of an extension from multiple phones. Business have needed (and implemented) that for years. Having such an implementation in asterisk would definitely be a major plus (regardless of what our definitions of a pbx and keysystem happen to be). Many people seem to want this feature. I think they are just confused. I've never actually heard of a good reason to let multiple devices register with the same username/secret. Most of the time they want a call to ring on multiple devices and they are trying to make a device == extension, which is not correct. A device is a device and an extension is an extension and they are not the same thing and there is no 1-to-1 mapping between them. Victor Alvarez wrote: I really think this matter deserves attention. I have been asked many times about it. Regards, Victor. Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I wonder if there is another way to do this. Of course, I am talking about a SIP proxy behaviour, simultaneous registration, both phones ringing at the same time and first to answer gets the call. Kind regards, Victor. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Full T38 sip Faxing now Available
Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: So you all think VoIP sypply is warm and fuzzy
Real scary who Don't bash my company. You've never used us. I test products for deployment so my customers don't have to call in for help. When I have to waste my time hunting down software then I get mad. Mediatrix has been great about getting this software to me. There is a limited web server built in to their products but you can not provision it without their software. And the person who said this was not the correct forum to complain to, your right. I'm sorry to all of you for informing you about the business practices of a supplier. I believe now that someone removed the standard provising cd that is shipped with every Mediatrix product. So everyone, make sure you receive all that you've paid for. I'm ending this thread now. Mark Musone wrote: ..all i know is that if this guy is bitching about a non-existant CD and is unable to provision a simple VOIP device, then i would be terribly afraid of his companies technical ability for their actual VOIP service.. scary..a VOIP provider that can't even provide themselves... On 7/19/05, Jason Stewart [EMAIL PROTECTED] wrote: On 18/07/05 17:06 -0700, Michael D Schelin wrote: I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Obviously you have a misunderstanding. Why not assume that there is a misunderstanding, with voipsupply then work from there instead of dumping your anger out on all of us? I don't doubt that there is a CD or there was once a CD that shipped with the 2102, but - According to the Medaitrix Web Site... --- Copy and paste from mediatrix web site --- With the Mediatrix 2102, service providers get the product characteristics allowing them to successfully deploy residential IP telephony applications. The Mediatrix 2102 provides a web interface, giving users a convenient access to the unit for initial set-up. The Mediatrix 2102 can auto-provision by fetching its encrypted configuration file from a TFTP or HTTP server making installation transparent to end-users. To further facilitate deployments, factory loaded configurations are possible. Automatic firmware and configuration file downloads ensure that the 2102 is always up-to-date. --- end --- You are supposed to use a web interface for initial set up. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: So you all think VoIP sypply is warm andfuzzy
After an extensive conversation with Mediatrx 's sales department , I stand corrected and so does the salesman who spoke to me. My apologies to Voip Supply. I understand now you never knew about the CD. Garrett Smith wrote: I though I would post an update for everyone on what DOES and DOES NOT come with every Mediatrix product. Every Mediatrix product, EXCEPT the 2102 comes with a CD. The 2102 does not come with a CD. As per Michael in the Mediatrix Sales Department, the CD and provisioning tool is a separate part number that needs to be requested at the time of purchase, and ordered in addition to the unit. Because this is a service provider unit, Mediatrix and their distributors DO NOT include the CD and PROVISIONING tool unless it is ordered. Why? Because once you get one CD there is no need to keep receiving them, when ordering in bulk. So, if any of you plan to order the Mediatrix 2102 from anyone, make sure you ask for the CD at the time of purchase. I would like to thank Mr. Schelin for bringing this to our attention and we will be updating our site to prevent this sort of problem from occurring again. If anyone has any further questions about the 2102, or how to order one with or without the CD, please contact me offlist. Thanks! Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael D Schelin Sent: Tuesday, July 19, 2005 1:35 PM To: Mark Musone; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: So you all think VoIP sypply is warm andfuzzy Real scary who Don't bash my company. You've never used us. I test products for deployment so my customers don't have to call in for help. When I have to waste my time hunting down software then I get mad. Mediatrix has been great about getting this software to me. There is a limited web server built in to their products but you can not provision it without their software. And the person who said this was not the correct forum to complain to, your right. I'm sorry to all of you for informing you about the business practices of a supplier. I believe now that someone removed the standard provising cd that is shipped with every Mediatrix product. So everyone, make sure you receive all that you've paid for. I'm ending this thread now. Mark Musone wrote: ..all i know is that if this guy is bitching about a non-existant CD and is unable to provision a simple VOIP device, then i would be terribly afraid of his companies technical ability for their actual VOIP service.. scary..a VOIP provider that can't even provide themselves... On 7/19/05, Jason Stewart [EMAIL PROTECTED] wrote: On 18/07/05 17:06 -0700, Michael D Schelin wrote: I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Obviously you have a misunderstanding. Why not assume that there is a misunderstanding, with voipsupply then work from there instead of dumping your anger out on all of us? I don't doubt that there is a CD or there was once a CD that shipped with the 2102, but - According to the Medaitrix Web Site... --- Copy and paste from mediatrix web site --- With the Mediatrix 2102, service providers get the product characteristics allowing them to successfully deploy residential IP telephony applications. The Mediatrix 2102 provides a web
[Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I don't here back from you shortly and receive the configuration program that should have shipped, I will return it back to you for a full refund and express my views to the Voip community. As of now I've herd of nothing but good things about your customer support. I've called and left messages to your support team. I waited 7 days for this unit and have no way to configure it. Email me the CD. Michael D. Schelin Owner Shelltel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
I was waiting for everyone to reply so here is mine.. Check out the Mediatrix web site. There are no downloads or lists of resellers who might have this provisioning software that is normally included with purchase. You may be right that it is a refurb but every indication points that it is not. I have contacted both companies and I'm waiting for replys. I'm on the west coast and it took over 7 days to get here. I am a little pissed when all other ATA's are configurable from their built in web server. And Yes, I'm self serving as well as mostly everybody I've ran into in this business. This unit was purchased for testing. Because of the timezone problem, When I get the product from UPS it's too late to call Canada or FL. when all I need is a simple download to correct the problem. Is it too much to expect everything in the box when you purchase it? Or have a web site with these free included software so if this happens we don't wast our valuable time. By the way I did get an email from VOip Supply asking me to wait until morning so they could find the software. This is at 2:30 PST. This complaint was to hear from others about VoIP Supply and their business practices. I wanted to get feedback ether way, or maybe a contact name so I can get this paper weight working and tested. Has anyone used the 2102? Please let me know. Michael D. Schelin Shelltel JD Austin wrote: Michael D Schelin wrote: Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I don't here back from you shortly and receive the configuration program that should have shipped, I will return it back to you for a full refund and express my views to the Voip community. As of now I've herd of nothing but good things about your customer support. I've called and left messages to your support team. I waited 7 days for this unit and have no way to configure it. Email me the CD. Michael D. Schelin Owner Shelltel Are you sure you didn't buy a refurbished model? I hear they sell a lot of refurbished equiptment, I've purchased some of it myself. Everything I've purchased from them worked without issue. None however came with an installation CD. A few things had to be reset to clear settings though. Anything I needed was freely available. Since you know how to contact Medatrix, perhaps you can download the software or get a CD from them. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] InfoWeek Article on VOIP
I agree with you but not 100% with them. An IP to Ip call on an ATA flat out is better . Now don't even get me started about cellular. My Service dosen't drop calls in the middle of conversations. VoIP is a notch better than Cellular. Michael Graves wrote: Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, ATT, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability call quality. It is interesting that all of these retail providers use ATA type devices. I wonder how some of the stronger true ITSPs like Level3 or even Nufone, VOIPJet, etc would fare, especially with an all digital scheme...ie hard IP phones. My own sense is that my IP base calls are cleaner than my SBC lines. I accept that they're less reliable, but much of that I attribute to the fact that I'm no Linux guru and I use a retail DSL line as my IP access. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any way to authenticate SIP peers using SRV?
Proxy servers can do that. Brian Capouch wrote: A group which my school is part of wants to start using DNS SRV records to allow email-style dialing amongst members of the group. I have gotten the records in our zonefiles, and things work pretty much just fine. However, since the DNS server can only specify a host and port, there doesn't seem to be any way to authenticate the user coming in. Is that the case? Is there a fix? Thanks in advance for anyone who might be able to shed some light. I've been to the Wiki and list archives, btw. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Gui?
search for [EMAIL PROTECTED] It works well and is very easy to install for beginners like me. Michael Felder wrote: Can anybody recommend an Asterisk GUI to help a newbie confg ? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: [EMAIL PROTECTED] http://www.ITMedic.com.au Keeping your computer systems healthy. ure Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: G729 licencing with asterisk, how does it work ??
G279 on Asterisk works great. Jean-Louis curty wrote: thanks I 'll try ... :-) jl 2005/7/4, Jean-Louis curty [EMAIL PROTECTED]: Hi, I'd like to understand what should i do to use G729 codec in a legal way, how do I order licences ? to whom ? how do I install them on asterisk etc ? thanks in advance , jl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-ss7] Asterisk - ss7
I thought everyone should know this. Jorge, After reading your page in the http://voip-info.org/tiki-index.php?page=Asterisk+SS7 please advise Your U.S. customers that SS7 is not done the same way as in the rest of the world and the requirements are different. The U.S carrier's require 2 redundant links. I know this first hand because we run an SS7 network. CARDOSO Jorge Miguel wrote: http://voip-info.org/tiki-index.php?page=Asterisk+SS7 ___ Asterisk-SS7 mailing list Asterisk-SS7@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-ss7 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Provider Survey
Call Mike at ShellTel 626-276-9009 List Receiver wrote: Having used Broadvoice for a while with marginal service, I want to move on to another provider. So my question to the List is who is good? I know now one service is perfect but somebody out there has to be decent. Who have you guys had the best luck with? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK
Hello, I'm not sure about Asterisk and in band DTMF without careful reading, but i do know that most ATA's and soft phones all have in band capabilities if set. G729 may not pass in band DTMF correctly all the time,in fact it's very poor and this is the reason for out of band. I think from reading the rest of the comments on the this post that you may have to look closer at encryption to keep all eyes from sniffing out the pins. I understand why you wouldn't want to slow down VoIP any more than you have to but customer security is more important. Robert Webb wrote: On Fri, 24 Jun 2005 13:10:13 -0400 (EDT) [EMAIL PROTECTED] wrote: On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote: We are on a real world... Every cyber cafe has its own little hacker/cracker that is sniffing out... A simple ethereal capture could give me a bank pin number... It is REALLY trivial! And this is different exactly HOW with inband DTMF?? They can do the EXACT same thing! If you want security don't use VOIP unless it's encrypted and/or over a VPN. It's really that simple. Ok, point me on HOW may I get DTMF inband with ethereal. Andrew, I'm just looking for the most quality/security solution to use Asterisk with G729, ok?! I think this is good for all of us. Thanks. Denis. People, could you PLEASE check first as to who your respons is going to. This double posting that has started recently is getting VERY annoying. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tellabs Echo Canceller
That sure sounds like it's Analog trunks to me. I believe you will need a channel bank to go from T1 to 24 ds0's and another to go back to T1. I could be wrong but I don't think so. I don't think that's what you were really looking for as far as an echo canceller. [EMAIL PROTECTED] wrote: I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked Receive In and Send Out. The 2 connectors on the drop side are marked Send In and Receive Out. I will be connecting the echo canceller between a PRI and our asterisk box. We recieve our PRI on an RJ-45 jack, so I assume I will need to make a cable that connects the Receive Tip/Ring pair from a Cat5 and wire it to pins 1 26 on the Receive In connector and take the Send Tip/Ring pair and wire them to pins 126 on the Send Out. The same thing will need to be done for the 2 drop side amphenol connectors so that I can plug an RJ-45 connector in to our Asterisk box. Does anyone have a suggestion on the easiest way to do this? I will make my own cables/connectors if necessary, but I suspect there are already adapters out and I just don't know where to look. I'd love some suggestions on the best pieces to use if I do have to make it myself too. Thanks for any help or suggestions! Eric ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tellabs Echo Canceller
Sorry Guys if I look dumb on this with my post but I've never seen T1's come on in that way before. Just disregard my post. Andrew Kohlsmith wrote: On Friday 24 June 2005 16:17, [EMAIL PROTECTED] wrote: The shelf has 4 25-pair amphenol connectors. The two on the line side are marked Receive In and Send Out. The 2 connectors on the drop side are marked Send In and Receive Out. I will be connecting the echo canceller between a PRI and our asterisk box. We recieve our PRI on an RJ-45 jack, so I assume I will need to make a cable that connects the Receive Tip/Ring pair from a Cat5 and wire it to pins 1 26 on the Receive In connector and take the Send Tip/Ring pair and wire them to pins 126 on the Send Out. The same thing will need to be done for the 2 drop side amphenol connectors so that I can plug an RJ-45 connector in to our Asterisk box. It's easy to connect: Network rx pair goes to your send out. Network tx pair goes to Receive in -- similarly your Asterisk-end rx pair goes to Receive out and your asterisk tx pair goes to Send in. I am guessing this shelf can handle 24 echo cancellation cards? It seems a little odd to split the T1s up across 4 connectors but if you're terminating to BIX or something it really does make sense. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show
I can do that. Please contact me off th email net. 626-814-2354 Michael D. Schelin - ShellTel Lee Barken wrote: hi Leon, We are initially looking for US only, but eventually would like to add international toll free numbers. We would like inbound IAX2 or SIP. Thanks, -Lee On Wed, 22 Jun 2005, Leon Sun wrote: What kind of toll free do you need? For US only or whole North America? Do you need carrier send incoming call to your Asterisk by SIP or by T1/E1 from Digium card? Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Barken Sent: June 21, 2005 6:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show Dear Asterisk Community, Does your company provide inbound 800# origination? If so, please read this message and e-mail us a quote for monthly co-lo hosting of our asterisk server and per-minute inbound 800# origination. The Prostate Cancer Research and Education Foundation (PC-REF) is a non-profit organization dedicated to helping prostate cancer sufferers and their loved ones. We have created a weekly call-in show using Asterisk that we offer as a FREE service to the public. Callers can ask their questions from world reknown experts, or just listen in. It's kind of like a talk show except you use your telephone, instead of a radio. We need a provider who can host our Asterisk Server and provide reliable IAX2 or SIP inbound 800# traffic. The show is one hour per week. We need the capability to support 100+ simultaneous callers. Most callers listen for the entire duration of the show. We have been working with another provider for the last several months, however, after many trials and tribulations, they have determined that their maximum capacity is 15 simultaneous callers. They will remain anonymous for the time being, as I truly believe that they worked very hard and to the best of their abilities. However, they were just technically unable to deliver to our requirements, despite their promises and best efforts. As they have been kind enough to offer a complete refund, I see no reason to embarass them in this forum. Therefore, it would be helpful (but not an absolute requirement) for your company to be able to port/migrate our 800 number so that we can keep our existing phone number. We are ready to move quickly and eager to establish a long term, mutually beneficial working relationship. This call in show has the potential to help many Prostate Cancer sufferers! Your assistance will be recognized and appreciated!! Many Thanks, -Lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Your also not in the U.S. Out here in Southern California it's $500.00 - $600.00 a month for T1's. Filippo Carone wrote: * Barton Fisher ([EMAIL PROTECTED]) ha scritto: I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? when i read so high prices for bandwidth i wonder why i get 10Mbps over optical fiber for 70Euros/month. and i'm not a business customer... fc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo problem
Hi Martin, There was an great post last week about echo. It stated that the order of the lines matters. It does. The channels must be listed last for the echo cancel and most other things to work. Rx and TX gain is one of the things also affected. Now I'm using TE110 card in my system. I hope this helps because I'm not sure about Analog lines. Martin Roy wrote: Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo problem
I'm sorry all, lines means config lines of code. Michael D Schelin wrote: Hi Martin, There was an great post last week about echo. It stated that the order of the lines matters. It does. The channels must be listed last for the echo cancel and most other things to work. Rx and TX gain is one of the things also affected. Now I'm using TE110 card in my system. I hope this helps because I'm not sure about Analog lines. Martin Roy wrote: Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pricing for DS3000P
Are you kidding! $4000.00 is cheap for a ds3 board! Even if you don't use all of the 28 t1's it's better because you will now be able to put in as many T1's as you will most likely need. Expansion will be just simple configuration change. Also as I've read in these forums, the interrupt issue should go away as this should only need 1. Don't let the term DS3 scare you. I have herd there are DS3 to T1 adapters out on the market for as little as $500. If you need more than 1 4 port T1 card you should buy the DS3 card unless of course you only need 5 T1 ports. Jay Milk wrote: What's so special about two tons of steel and a little plastic and leather that you'd pay at least $20K for it? How come Adobe gets away with charging $300 for a simple CD, when you can buy a stack of 100 for less than $20? Content matters... And someone needs to pay for the development cost, testing, certification, etc... Or there wouldn't be any peripherals. -Original Message- From: Tom Fanning [mailto:[EMAIL PROTECTED] Sent: Saturday, June 04, 2005 3:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Pricing for DS3000P Agreed, those are the figures we were able to get from Digium... I'm still waiting for a confirmation, but I'm being safe with a $4k estimate.. snip What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to ensure that software echo cancellation ison?
Thanks Rich, seems other things are now working for me as well. good FYI! Rich Adamson wrote: Problem solved. this zapata.conf works (i.e echo is gone, echo cancellation is detected on zap show channel): context=from-pstn switchtype = national signalling = pri_cpe echocancel=yes echotraining=500 echocancelwhenbridged=yes faxdetect=incoming group = 0 channel = 1-8 this zapata.conf has echo cancellation not work: context=from-pstn switchtype = national signalling = pri_cpe group = 0 channel = 1-8 echocancel=yes echotraining=500 echocancelwhenbridged=yes And if I add faxdetect=incoming as the last line in that one, asterisk doesn't even start. I get the Ouch message and asterisk crashes. Lesson learned: the order in which lines appear in zapata.conf does matter. Questions to be asked: is that expected behavior? is it documented somewhere? Yes. For the most part, the order of the statements does not make any difference. However, what you've learned from above is not the order at all; its that your echo statements were not stated before the channel = 1-8 statement. In other words, had those echo statements been included prior to the channel statement (as in your working cofig), the order within the channel definition section is irrelavent. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?
Just asking the forum community - Is there an advantage in changing the syntax? Steve wrote: I stink at regular expressions, but can always find what I need to get a job done using google :-) Don't use vi (unless you figure out how to do it in vi). I won't be much help there. in this case sed is your friend. It's a breeze to use too. Here's what it looks like: sed -e s/text_to_find/text_to_replace/g inputfile outputfile and yup you can use the same name for both files to simply update the file. Make sure you back it up first of course. Here's a link with some more examples using sed: http://pegasus.rutgers.edu/~elflord/unix/sed.html Hope this helps! Steve On Sat, 4 Jun 2005 [EMAIL PROTECTED] wrote: Finally I decided to rewrite my dialplans according to the right sintax, that is exten = someexten,priority,application(arg1,arg2,...) should be exten = someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex __ TISCALI ADSL 1.25 MEGA a soli 19.95 euro/mese Solo con Tiscali Adsl navighi senza limiti di tempo a meno di 20 euro al mese e in piu' telefoni senza pagare il canone Telecom. Scopri come http://abbonati.tiscali.it/adsl/sa/1e25flat_tc/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 vs. gsm
I have used G729 and it sounds almost as good as G711U. The problem is the way Asterisk uses it. It does not sound robotic and it's not suppose to sound that way. Most Carriers want the calls to be in g711u so thats why I use G711u otherwise I want to save money on bandwidth. G729 on Asterisk adds latency. this could be one of your problems. Also you will not get music on hold to play well with G729. Andrew Kohlsmith wrote: On May 27, 2005 06:12 am, chawki hammoud wrote: I installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? It sounded more or less the same to me, perhaps with GSM being a little more human (I can easily listen to music on hold with GSM). I am interested in G729 because the internet in my country is very expensive and I want to save every bit possible. I want to use G729 because it takes less bandwidth for each additional call between two IAX servers than other codecs. Make sure you use IAX2 trunking then. It can give you very large bandwidth savings when you have multiple audio streams between two servers since the UDP overhead is not repeated for every call. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 vs. gsm
Steve, you should really test the Codec and have G729 running as a pure IP to IP call you can not hear the difference on good networks! Please do not get me wrong that G711u sounds better through the PSTN. Thats a given! You can't convert G729 up and down to G711 and expect the sound quality to be there. I'm a carrier and have tested G711 and G729 and have found that they both sound great through dedicated hardware. Asterisk's colors the G729 a little. Also my hearing is fine. Please do not put down the comments of others in this forum. I'm stating my comments from my real world trials and this is not bad information. The man must compare codecs on his own and see what works for him. For me we've stuck with G711u because it's best through the PSTN. If I was running a pure IP to IP system I would use G729, Iblc, or GSM. Mike Steve Underwood wrote: Michael D Schelin wrote: I have used G729 and it sounds almost as good as G711U. The problem is the way Asterisk uses it. It does not sound robotic and it's not suppose to sound that way. Most Carriers want the calls to be in g711u so thats why I use G711u otherwise I want to save money on bandwidth. G729 on Asterisk adds latency. this could be one of your problems. Also you will not get music on hold to play well with G729. G.729 doesn't sound that bad. However, if you find it hard to tell G.729 from G.711, I think you should have your hearing checked. :-) It really doesn't help people to assess what is right for them, if other people make these exaggerated and unreasonable claims. Even the people promoting G.729 give it a MOS far below G.711. You are certainly right about music on hold, but even voice plus a little background noise can sound bloody awful through G.729. Its performance is *very* dependant on compressing just a single human voice. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 vs. gsm
Clue or clueless? Your call. Steve Underwood wrote: Michael D Schelin wrote: Steve, you should really test the Codec and have G729 running as a pure IP to IP call you can not hear the difference on good networks! Well, it does to anyone without hearing damage. It sounds very obviously different. Please do not get me wrong that G711u sounds better through the PSTN. Thats a given! You can't convert G729 up and down to G711 and expect the sound quality to be there. I'm a carrier and have tested G711 and G729 and have found that they both sound great through dedicated hardware. This is meaningless drivel. Asterisk's colors the G729 a little. Also my The only time when Asterisk colours G.729 is when there is packet loss. Asterisk isn't handling that well. hearing is fine. Please do not put down the comments of others in this forum. I'm stating my comments from my real world trials and this is not bad information. Since it doesn't correlate with the impression of even the developers of G.729, it *is* bad information. Realistic people know G.729 will be worse. What they need is meaningful guidance as to just how much. The man must compare codecs on his own and see what works for him. For me we've stuck with G711u because it's best through the PSTN. If I was running a pure IP to IP system I would use G729, Iblc, or GSM. In a sane world pure IP to IP systems would't use G.711, G,729, iLBC, or GSM. They would be usign a wideband codec, as Skype does. Look at the favourable impression people have of that. Next time, its probably better to argue with someone who hasn't spent time in speech codec development. We tend to have a clue what we are talking about. :-) Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help accepted :-)
Hello, I too am having an echo issues. My partner an I have discussed this in depth and believe that digital circuits can not create the echo problem. It's when it hits the Analog network or in my cases ATA's that are having echo problems. I have another gateway that does not have any echo on the same Sipura ATA's. I have also notices that the Asterisk levels are very hot from T1-PRI to Sip. I have not had the chance to turn them down but I think this is where my echo problem is occurring because the Digital levels are too much for the ATA to convert. I will let you know if I find this to be true. Michael Schelin / ShellTel Ronald Hartmann wrote: Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123- the call sounds great no echo what so ever. If the person at 123- hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party does not notice any difference. I have tried all the following. #define CONFIG_ZAPTEL_MMX tried this defined and undefined Then tried each of the following types of echo cancellations. #define ECHO_CAN_MARK #define ECHO_CAN_MARK2 with and without #define AGGRESSIVE_SUPPRESSOR #define ECHO_CAN_MARK3 I am completely at a loss on how to get rid of this echo problem. The system is completely useless for incoming calls, as it currently stands. Is there a Digium card that handles echo better? Are there any asterisk compatible cards with hardware echo cancellation available? Thanks Ron [Zapata.conf] span=1,1,0,esf,b8zs bchan=19-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us [Zaptel.conf] [channels] language=en signalling=pri_cpe switchtype=national pridialplan=unknown echocancel=yes echocancelwhenbridged=yes (tried no) echotraining=400 (tried 800 also) usecallerid=yes callerid=asreceived overlapdial=yes immediate=no group=0,1 context=from-pstn channel = 19-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help acc epted :-)
I have found that the audio is hot from some carriers and low on others. I have found that this is causing the echocanclers problems. Before I reduce it down by 3db I will see if some of the problem in in the Supura . Andrew Kohlsmith wrote: On May 26, 2005 01:58 pm, Colin Anderson wrote: I have had good success fiddling with the txgain and rxgain values in zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated most of the echo, and training the users to turn down the gain on their handsets did the rest. It's true, with a PRI, that gains are cranked across the board. Turning the gain down solves a lot of echo problems, with negligible effect on voice quality. There should be *NO* reason to adjust tx/rxgain on a PRI or ANY digital connection! The fact that adjusting it down 10% worked suggests that the telco switch is boosting the signal for some unknown reason. Use the technique outlined in this message: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html (It works VERY well and is very straightforward, thank you Kris, I reference this all the time!) Again if you're screwing with gains on PRI you have bigger problems, I think. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] origination providers
I can give you all the simulataneous calls you need for $.02 / min. in the U.S. and Canada. Please call me at 626-814-2354. Michael Schelin Shelltel Kanuri, Seshu (Company IT) wrote: Mike, Many of the providers I've tried contacting either won't call me back, or want me to sign an NDA just to get a rate quote, or some other bullshit. Assuming that you will need about 12 to 24 simulataneous calls on each DID you want to run, and you are using Ulaw to get these calls, what is the bandwidth that the DID provider has to give you, apart from the DID service? Ulaw needing 64 kbps per line, needs 1.2 mbps for 20 simultaneous calls. Assuming a Data T1 costs about $500 bucks a month and assuming that you need/use the DID for only 8 hours a day at that rate, it costs about $100 per month in data bandwidth alone. Who will pay for this, If it is not Democracynow who is footing the bill? Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of mike castleman Sent: Tuesday, May 24, 2005 3:00 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] origination providers hi folks, Has anyone found a good (and, ideally, cheap -- we don't really want any per-minute charges) origination provider which can handle a moderate number of simultaneous incoming calls (to the same, single DID)? Many of the providers I've tried contacting either won't call me back, or want me to sign an NDA just to get a rate quote, or some other bullshit. Most of the providers whose rates are plainly posted on their website have a limit of at most 4 or 6 simultaneous calls, which is not likely to be enough for the application I'm considering. You can reply off-list or on-list, as you prefer. many thanks, mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (mobile) NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] origination providers
Mike - If you don't mind Los Angeles Area DID's then I can supply you with Fixed costs with no per minute charges on your inbound calls. If This is what We sell. Please call Michael Schelin at Shelltel 626-814-2354. Ed Greenberg wrote: Hi Mike, Understand that your supplier will be paying by the minute. What you want is your suppliers worst nightmare. Fixed income and variable (increasing) costs, both to his upstream provider and also in bandwidth, both network and computer. Many of us will be happy to supply you with as many simultaneous calls as you can handle, but we'd all want to be paid in proportion to your usage, or we'd be out of business. Best, /edg --On Tuesday, May 24, 2005 4:10 PM -0400 mike castleman [EMAIL PROTECTED] wrote: I'm not entirely sure what you're asking. The application in question will involve setting up asterisk in a datacenter where we already have a fair amount of bandwidth. As far as the DID provider's portion of the bandwidth, I assume that they would account for this in the rate they quote us. I'm just asking the list if they have any good experiences with origination providers, as my attempts to get them just to return my calls have not been successful. If it's relevant, I imagine gsm or speex codec for this application, but haven't yet made a decision. mike On Tue, May 24, 2005 at 03:42:43PM -0400, Kanuri, Seshu (Company IT) wrote: Assuming that you will need about 12 to 24 simulataneous calls on each DID you want to run, and you are using Ulaw to get these calls, what is the bandwidth that the DID provider has to give you, apart from the DID service? Ulaw needing 64 kbps per line, needs 1.2 mbps for 20 simultaneous calls. Assuming a Data T1 costs about $500 bucks a month and assuming that you need/use the DID for only 8 hours a day at that rate, it costs about $100 per month in data bandwidth alone. Who will pay for this, If it is not Democracynow who is footing the bill? Seshu -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] origination providers
Hello All, After a wonderful conversation with Mr. O'Shield he asked me to inform the Asterisk community that we run an SS7 network with Los Angeles Area trunks and DID's that we can supply with Fixed costs and NO per minute charges on our inbound trunks. Calls include caller ID. Also we can supply Co-Location, Power, and a fixed amount of non shared Internet bandwidth. This can be used for Calling card providers or call centers. We are tied onto the Verizon Backbone. Please call Michael Schelin at ShellTel 626-814-2354 for more information. Michael D Schelin wrote: Mike - If you don't mind Los Angeles Area DID's then I can supply you with Fixed costs with no per minute charges on your inbound calls. If This is what We sell. Please call Michael Schelin at Shelltel 626-814-2354. Ed Greenberg wrote: Hi Mike, Understand that your supplier will be paying by the minute. What you want is your suppliers worst nightmare. Fixed income and variable (increasing) costs, both to his upstream provider and also in bandwidth, both network and computer. Many of us will be happy to supply you with as many simultaneous calls as you can handle, but we'd all want to be paid in proportion to your usage, or we'd be out of business. Best, /edg --On Tuesday, May 24, 2005 4:10 PM -0400 mike castleman [EMAIL PROTECTED] wrote: I'm not entirely sure what you're asking. The application in question will involve setting up asterisk in a datacenter where we already have a fair amount of bandwidth. As far as the DID provider's portion of the bandwidth, I assume that they would account for this in the rate they quote us. I'm just asking the list if they have any good experiences with origination providers, as my attempts to get them just to return my calls have not been successful. If it's relevant, I imagine gsm or speex codec for this application, but haven't yet made a decision. mike On Tue, May 24, 2005 at 03:42:43PM -0400, Kanuri, Seshu (Company IT) wrote: Assuming that you will need about 12 to 24 simulataneous calls on each DID you want to run, and you are using Ulaw to get these calls, what is the bandwidth that the DID provider has to give you, apart from the DID service? Ulaw needing 64 kbps per line, needs 1.2 mbps for 20 simultaneous calls. Assuming a Data T1 costs about $500 bucks a month and assuming that you need/use the DID for only 8 hours a day at that rate, it costs about $100 per month in data bandwidth alone. Who will pay for this, If it is not Democracynow who is footing the bill? Seshu -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play gsm files in windows
The standard Windows recorder will play GSM files. You must make sure you set the correct values. Codec, playback rate, etc. Walt Reed wrote: On Mon, May 23, 2005 at 03:34:44PM +0100, Brett, Gary said: Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. See the Wiki: http://www.voip-info.org/wiki-Asterisk+sound+files ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P without router ???
I'm using it and it's great. I have it doing very basic routing from the PSTN to SIP. Manjit Riat wrote: Hi, I was going to order the T100P but it is replaced by TE110P. On further reading the TE110P does not need an external router (The one that separates the data from pstn lines ?). Has anyone got it configured? And on the wiki it says that the drivers for some distros don't exist yet. Is redhat supported? And if I need to connect a fax machine will the FXO cards on ebay work (I know to connect a fax a FXS card is needed) or will I need to get a TDM card (with FXS module) or an ATA. I am trying to stay away from ATA because I know there are some problems with faxes as faxes are really picky. So having an ATA means signal from PRI gets converted to SIP at asterisk - ATA converts SIP signal back to analog and then sent to fax. Will that work flawlessly or should I be on the safe side and get a TDM card so that no VoIP conversion takes place. thanx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
I think that it's fixed now with the V6 beta code. Anton Krall wrote: That's what I was starting to think.. Since I've always used ulaw or alaw... Seems that firmware 1.0.5.23 has ilbc broken. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin McCauley |Sent: Jueves, 19 de Mayo de 2005 10:15 a.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc | |Anton Krall akrall-lists at intruder.com.mx writes: | | | Guys, anybody having problem with ilbc and GS ata 286? I |just tried it | for fun (always using alaw) and voices sounded quite bad... crappy | voice prompts, not bad quality, just like weird noises. | | Anybody had this? whats the latest FW for those units? | | ___ | Asterisk-Users mailing list | Asterisk-Users at lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | | |Anton, | |I use iLBC exclusively on the 286/486 and it interoperates |with other devices on my network fine. In fact I use iLBC |because some of the people I talk to only have dialup and it |works the best for that. | |I will mention though, that I have stayed on FW version |1.0.5.16 since I have had troubles with newer versions. | |-Kevin | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
TCP is too slow for Real time Apps. If you have packet errors TCP will try to resend the packet. This will create latency issues. This is why UDP is used for Voip. 1 or 2 missing packets is not going to be missed. If you look at your Stats. you'll see a few of them. Stewart Nelson wrote: I am interested in implementing RTP over TCP Why? If you want to permit operation through a firewall that blocks UDP, there are packages that provide VPN tunnels over TCP or even HTTP. You could then run any VoIP system over that VPN. As you said, delay performance would sometimes be awful. Skype will automatically fall back to TCP if a UDP connection attempt fails. Most of the commercial instant messaging packages that support voice or video can work over TCP. If your purpose is to improve performance on networks with high packet loss rates, IMHO you would get better results from a UDP-based system that permits forward error correction, by transmitting each voice frame in two or more packets. If you can't afford the increased bandwidth, a system of retransmission such as used by popular streaming protocols would still be better than TCP. One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? Any of the above systems can work through NAT. If both endpoints are behind NATs, and you can't set up port forwarding on either, then of course you must connect via an intermediate server. Skype and the IM services do that automatically. If your desire for TCP is not related to firewalls or packet loss, I'd be interested in hearing about your application. --Stewart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Satellite Providers
The delay in the air is minor. Radio travels very fast through the air. Almost at the speed of light. It's the electronics that are causing the delays. The less electronics touching your signal the better. The up and down is very fast. But then you have all the converts and the land line links to factor in. Microwave also has delays such as the Motorola equipment which is only half duplex. This will also incress the time. Max is right, check into some ground based systems. Max W Blackmer Jr wrote: Satellite delays are always bad. It is more a delay because of the time it takes a signal to travel to the satellite and back to a receiving station. You might want to check into ground station to station microwave communications stations. The best is to have a tap to a phone company that may have cell towers in the area. Cheers, Max Original Message Subject: [Asterisk-Users] Satellite Providers From: "Yiannis Costopoulos" [EMAIL PROTECTED] Date: Wed, May 11, 2005 12:23 pm To: asterisk-users@lists.digium.com Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream firmware 1.0.6.2
this is beta code! I'm beta testing The t38. Don't use this unless your testing. It is not backwards compatible. Julio Arruda wrote: Doug Lytle wrote: Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ 2 quick notes, a quick test seem to indicate iLBC is broken (didn't try any troubleshooting). And, in the release notes, from what I remember, there are mentions of problems with dowgrading it, at least they recomend you to call support to do it) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Who's happy with their voip service?
Please Give me a call. I'm the owner of Shelltel. 626-814-2354 We're not the same cookie cutter VoIP carrier. Bryce W Nesbitt wrote: I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change Who's happy with their voip service using asterisk? Where do you get reliable DIDs? The 'carrier partner' they speak of.. can you get the did directly from them? Are all the voip providers this flakey? I've tried 5 providers, and I can't say that I'm happy with any of them. I'm far to small to deal with 'carrier partners' directly (e.g. Level 3, XO or RNK). So I have to deal through resellers. And they all seem to be operating on shoestrings and duct tape. I'm OK with the awkward setup, confusing configuration, and (for Asterisk) all but useless documentation. But high latency, dropouts, unplanned outages, lack of clues, echos, all take the shine off things. With Asterisk I have very few tools to monitor connection quality, especially on the outbound leg of my calls. I at least want to know when it sucks, and have some control over parameters. I keep a POTS line at home. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who's happy with their voip service?
Isn't amazing what has happened in the last five or six years with the Internet. There is no design flaw with IPv4. It was created back when you were in diapers and with todays pda's having more power than the systems back then. An industry protocol that is going strong 30 or more years is amazing in it's own right. How could they see the future of what we are doing today with the protocol. I believe the engineers who designed IPv4 were brilliant men and did a great job designing something that is computer system neutral. Again in the the last few years VoIP has come a long way as the PSTN has had over 100 years to perfect theirs. If we did not have to interface with the PSTN don't you think we would be better off? They didn't have to interface with anybody else. Chris Coulthurst wrote: I tend to agree about the in-house being the 'stable part'. Like anything else on the internet, if you don't have control of all parts (trunks and phones and dialplans), there are bound to be issues with uptime, and how your equipment responds to 'their' downtime. It reminds me of the headaches I had as an ISP when a BGP4 route wouldn't switch to the redundant carrier, because the main carrier didn't really die, it just stopped transmitting! It's also worth noting the design flaws with IPv4 handling priority packets in the first place. I think most of the little 'gotchas' in VoIP would magically vanish if QoS was something that could be depended upon. All you need is one router to not know how to pass the qos token, and now you don't really have any! Its another example hiw an in-house system can be stable when you hold all the cards. By the time IPv6 gets here, it will be amazingly obsolete... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Andre Normandin |Sent: Saturday, May 07, 2005 8:00 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Who's happy with their voip service? | |I've had Broadvoice for over a year now, and although their outages are |really annoying, the fact that their service costs $20/month unlimited is |what keeps me with them.. | |I have 2 Inbound #'s through them (same account), one in GA (678-253) and |one in CT (203-935), and overall their inbound has been more reliable than |their outbound (minus the past week or so).. | |I have my dialplan try BV first, and then if it cannot use BV for outbound, |it rolls to my pots line(s).. It actually works really well, except that if |BV goes completely toes up Asterisk decides that it doesn't want to do |anything either :-( | |That is what I find the most annoying, quite frankly, BV is having Growing |pains (in my opinion), and I can accept that, haven't put anything critical |on my BV inbound, and 90% of the time BV outbound works fine.. The rest of |the time, the pots take care of outbound, and anyone who calls me calls on |my pots lines (except for family in GA, which is why I have the GA #). | |For me personally, I just think VOIP is 'too' early in the maturity curve |to |really rely on it as a provider.. It's great in-house (medium/large |companies), but for service, I think pots are the way to have rock solid |service for the time being. | |I know of two of my friends that have Vonage as their only inbound numbers |(not via asterisk, via the vonage locked adapters, so it is completely |vonange), and their service also has issues at times.. Granted, I'm not |sure |if it's a true vonage issue, or their internet connection, but nonetheless, |there are still issues.. | |If I could get Asterisk so it just work continue 'working' properly with |whatever SIP connections it can reach, I'd be a happy man.. Don't get me |wrong, I think Broadvoice needs to communicate better with their customer |base, and the latest ongoing outage is, to say the least, very frustrating, |but I am willing to cut them some slack because I think VOIP is still in |it's infancy, and broadvoice is the only BYOD provider I know that will |give |unlimited for $20.00/month.. | | - Andre | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] Behalf Of Johnathan |Corgan |Sent: Saturday, May 07, 2005 12:58 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Who's happy with their voip service? | | |JD wrote: | | Inbound | calling has been down for 2 days. | |Just FYI, mine is back up (408-903) as of about five hours ago. | |I did just speak with a (Broadvoice) support tech on an entirely |unrelated matter (40 min. hold time!), mentioned mine was working, and |he seemed to think things were coming back in stages. | |I've had them for two months now. People may recall a series of emails |regarding packet loss through their PNAP link to Sprintlink (my ex-ISP |backbone.) I ditched the Sprint BBD fixed-wireless service, got |Sonic.net DSL, and have been enjoying
Re: [Asterisk-Users] Broadvoice Issues
Hi Guys, give me a try. I'm Michael Schelin of ShellTel and we are a business Voip service provider. I have very little down time and we work 100% with Asterisk. Please call 626-814-2354 or email me [EMAIL PROTECTED]. I'm a little more the the discounters but when you need help I'm there! No hold for hours and we own our own network so we have wholesale services like origination and termination. Thanks Rich Adamson wrote: Someone in another thread suggested that BroadVoice reads this forum -- I hope so. I am a prospective customer, only inasmuch as their advertised rates are so attractive. But as a consultant it will be a cold day in the Ether before I recommend them to a client of mine until these "issues" are cleaned up. It's difficult, very difficult, for a newcomer to this technology to figure out who is a reliable provider and whether companies like BroadVoice are a flash-in-the-pan, here today - gone tomorrow, take your money and run, outfit or whether they are just having growing pains. The repeated outages are unacceptable. If they are planned then they should have the professionalism to send every customer an e-mail advising of the outage BEFORE the outage. They should also put a web page up notifying customer of planned outages, loads on the proxies (latencies) and other information a business who was dependent upon their services would need. Finally, to BroadVoice, I say, if you want my business then clean up your act and do so promptly. Otherwise the VoIP community will just dismiss you as another provider wannabe. Those of us that have been around the list (and BV) for awhile know that one of the BV employees had a strong interest in making * work with their service. He did so and does frequent this list. However, he's a one-man support shop (with BV clearly stating there is no BV support for *). So the * users use this list for support. If you are going to act as any reasonable consultant or reseller, it certainly is not in your client's best interest for you to recommend BV when it is simply unsupported and you know it. Until BV as a company steps up to the plate, it serves no useful purpose to bitch at them. Their doing exactly what they said they would do... nothing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: Call forwarding]
As far as I know Asterisk does not support normal PSTN type call forwarding. I.E. the user would type *72 etc. This is called call forking. My Mulitech gateway does but at a huge price. Also T38 is supported. I have several carriers that I use that have Asterisk. All of the Asterisk boxs won't accept call fowarding. I send the calls to my carriers with Cisco gateways and the calls reroute correctly. Now I have a proxie that controls everything. You may be able to do call fowarding with 2 boxes. But a call in and reroute back out may not work. Damian Funnell wrote: Any takers? Sometimes the most basic questions yield the least replies, huh? Cheers, Damian. Original Message Subject: Call forwarding Date: Wed, 04 May 2005 08:40:41 +1200 From: Damian Funnell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com References: [EMAIL PROTECTED] [EMAIL PROTECTED] Hi team, Basic question I know, but I can't seem to find any obvious information about this: Does anyone know if * natively supports call forwarding from a given extension (i.e. call forwarding without having to write a macro)? My user wants to be able to dial a code plus a phone number to start diverting all calls to the given extension to that number. Call forwarding would then be disabled by dialling a code number again. I expected that * would support this type of feature natively, but can't find anything in the wiki. If responding please let me know if we need to enable anything in features.conf as well. Thanks in advance, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail Greetings
Hi all, What would cause the greetings not to play. The u command is
supposed to play the unavailable greeting. It doesn't work. with this
setup. Maybe I'm missing something. The voice prompts play well. What
do you think? Thanks
exten = 9007,1,VoicemailMain
exten = _.,2,Voicemail(u${EXTEN})
exten = _.,2,AbsoluteTimeout(180)
exten = _.,4,Congestion
exten = _.,5,Hangup
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Re: [Asterisk-Users] TE410P on Dell 2650
Have you turned off all the unused I/O ports, IE: serial, USB, Printer? Aza wrote: I have a problem with a Dell 1850 and a TE410P card as do a few others who posted over the weekend. The problem in this case isn't so much echo but static and chop on all calls using ZAP channels. My zttest results look pretty much the same as yours. We were thinking it was the RAID controller but someone posted that they had the same problem with an 1850 without the RAID controller. We've got some E100P cards we're going to try out but it's difficult to troubleshoot this issue when it involves live PRIs. Aaron From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charlie Watts Sent: 04 May 2005 20:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P on Dell 2650 I have a Poweredge 2650 w/ a TE410P card. I'm getting lots of local echo (my users hear themselves) when calling local telephone numbers. Echo cancelling helps, but doesn't solve the problem. From zapata.conf: echocancel = 64 echocancelwhenbridged = yes echotraining = 800 My zttest results consistently look like this. The digium fellow who has been helping me says these are lower than he'd like to see. [EMAIL PROTECTED] zaptel]# ./zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% --- Results after 9 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.989149 The card isn't sharing interrupts, etc. We're at the point where the Digium fellow is suggesting other motherboards and avoiding SCSI. Anybody else have a 2650 with a TE410P ? Do you have any local echo problems? What do your zttest results look like? Are you satisfied with the system? Thank you very much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail Greetings
I have fixed and rearranged the priority and still no client
greetings. The commands below have been fixed but all I get is the
system prompts. That is ok but my clients should be able to have there
recorded greetings played on VM access. I can record the greetings and
play them back but just not when a sip call is accessed. what is
going on. When debugging the sip call there is nothing stated about
playing the client s greeting.
snacktime wrote:
On 5/4/05, Michael D Schelin [EMAIL PROTECTED] wrote:
Hi all, What would cause the greetings not to play. The u command is
supposed to play the unavailable greeting. It doesn't work. with this
setup. Maybe I'm missing something. The voice prompts play well. What
do you think? Thanks
exten = 9007,1,VoicemailMain
exten = _.,2,Voicemail(u${EXTEN})
exten = _.,2,AbsoluteTimeout(180)
exten = _.,4,Congestion
exten = _.,5,Hangup
You have two priority '2' extensions and you are missing priority '1'
and '3'. I think that extension would just timeout, but I've never
tried that particular setup myself:)
Chris
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Re: [Asterisk-Users] Collect calls
You Bring up a great point. I understand these codes and my system brings them in via ss7 but as youself I don't know how to protect my network from these charges. I will follow this post to see if anybody has a fix. Rodrigo P. Telles wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Folks, Does someone knows how to identify and block collect calls on Asterisk using PRI channels? I googled it and found this: http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html I don't know what does it mean!!! Can someone help me to understand this? I tried to apply that way too, using Flash() but Flash() complains and looks like just work with FXO channels: http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html Thanks in advance. - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCd+zUiLK8unYgEMQRAkChAJ4xDYOvl8yZY+Uqn6v5VFZ4tMzicQCfT8+T 5foewh0m/o3ABMqcNHhtQs4= =rsu2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do you get rid of Spawn's
Hi everybody, How do I get rid of spawn's ? example -- Executing Dial(Zap/23-1, sip/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Accepting call from 'xxx3672728' to 'xxx2769906' on channel 0/23, span 1 -- SIP/xxx.xxx.xxx.xxx-0adc is ringing -- SIP/xxx.xxx.xxx.xxx-0adc answered Zap/23-1 -- Channel 0/23, span 1 got hangup == Spawn extension (from-pstn, xxx2769906, 1) exited non-zero on 'Zap/23-1' -- Executing Dial(Zap/23-1, sip/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' -- Got SIP response 404 No user [EMAIL PROTECTED] at this server back from xxx.xxx.xxx.xxx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Amp extensions script
Hi, Is there a script in amp for adding the extensions? And can it be modified? When adding a new extension it rewrites all of the information it the context blowing out my additions. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern Matching
Hey Mojo, I'm thinking you might try using priorty 's to set some kind routing. just a thought.. Mojo Jojo wrote: We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in use/configured. I have been successful at setting up pattern matching as a catch all for 98 or so numbers not in use yet and I have been successful setting up the 2 numbers I want to make use of for now. Problem is, I can't use both at the same time! If I turn on the pattern matching then my greeting plays for the configured number, then the message plays for the invalid number (basically executing the extension with the pattern matching). I have read about sorting with pattern matching by using an include, I did this but it's not really helping. I have set a response timeout after the first extension plays it's greeting, I would think it should wait until it times out but it doesn't, it just immediately moves to the pattern matched extension. I must be missing something big here.. Any help is appreciated.. -- Private Label Wholesale Internet Access! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware Recommendation
I just read a great paper that said turn off anything that won't be used. Serial, USB , Printer ports, ETC. No Xwindows! Daniel Salama wrote: Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full capacity (96 simultaneous calls). This box will also deliver these calls to SIP users and record all their conversations via Monitor. I've heard that it's not necessarily a matter of memory (RAM) nor the need to have a multi-processor machine. But what really matters is that the motherboard (architecture) is designed to handle such a high amount of interrupts generated by the TE4XXP, the NIC, the storage array (whether it's SCSI or IDE or SATA). Does anyone have experience with particular brands of either motherboards they recommend are capable to handle this or complete systems (e.g. Dell or whichever brands), that are ready for this? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I add an IP to an Exten
This works from-pstn just fine. exten = 10 digit Inbound PhoneNum from the pstn,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,) How can I add the Variable exten with the proxie IP address. I want the exten to call my proxie. exten = _.,1,Dial(sip/[EMAIL PROTECTED] IP address,,r,) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Audio sent using playback cmd
Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux [EMAIL PROTECTED] version 0.9 CentOS release 3.4 (final) Linux 2.4.21-27.0.1.EL Hi All, I really need help on this. What would keep Asterisk from playing out audio files using the (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. In to Asterisk bun nothing coming out. Because I can hear the audio with the play tone I know there is something preventing the playback cmd from working. I had Audio Thanks _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: [Asterisk-Users] Voicemails stopping]
Chris, Try upgrading Lenux. I did mine with Yum Update and now I got voice. Chris Stinson wrote: Has anyone else had this issue? Original Message Subject: [Asterisk-Users] Voicemails stopping Date: Tue, 26 Apr 2005 13:04:55 -0500 From: Chris Stinson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Organization: ISDN-Net, Inc. To: asterisk-users@lists.digium.com Has anyone ever had an issue with a voicemail cutting off and then going to the menu, then by pressing 5 the voicemail will play a bit further then cut off again? After hitting 5 once more it will play the rest of the voicemail message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Audio sent using playback cmd
FYI To All, I fixed my problem by doing a Linux upgrade by typing Yum Update and taking the CentOS updates. My problem is solved. I have no clue what was going on but now I have Audio Now. Michael D Schelin wrote: Asterisk 1.0.7 built by [EMAIL PROTECTED] on a i686 running Linux [EMAIL PROTECTED] version 0.9 CentOS release 3.4 (final) Linux 2.4.21-27.0.1.EL Hi All, I really need help on this. What would keep Asterisk from playing out audio files using the (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. In to Asterisk bun nothing coming out. Because I can hear the audio with the play tone I know there is something preventing the playback cmd from working. I had Audio Thanks _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?
Tim, I found something in the huge log file. Apr 25 23:25:51 DEBUG[2330]: Ooh, format changed from unknown to ulaw what does this mean Ooh ? to 208.41.254.119:5060 Apr 25 23:25:51 VERBOSE[2330]: -- Executing [1;36;40mPlayback[0;37;40m([1;35;40mSIP/208.41.254.119-089b23d8[0;37;40m, [1;35;40mtelephone-in-your-pocket[0;37;40m) in new stack Apr 25 23:25:51 DEBUG[2330]: Ooh, format changed from unknown to ulaw Apr 25 23:25:51 DEBUG[2330]: Scheduling timer at 160 sample intervals Apr 25 23:25:51 VERBOSE[2330]: -- Playing 'telephone-in-your-pocket' (language 'en') Apr 25 23:25:51 VERBOSE[2330]: I see all log files. Sip debug is on so I see it wanting to play the file. I'm the root. I was thinking path but but I tested it by changing the name to something wrong. The call went right to and error tone. When It's correct it's just silence. The tones are not in the same directory as the sounds. Asterisk knows the directory. I will look at the log file. I have looked at the asterisk.conf docs. and I can't find the Variable need the to put in for the sound files path. Example below is my asterisk.conf file. I would like to force it to the correct directory. [directories] astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Tim Connolly wrote: Are you seeing anything in your /var/log/asterisk/messages file or even on the console with verbosity at 3 or more? I'm guessing you have a path or permissions problem, but you should see either in the logs or the console. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Monday, April 25, 2005 8:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Playback dosen't play Playtone(Congestion) does ? Hi All, What would keep Asterisk from playing out audio files (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. Because I can hear the audio with the play tone I know there is something preventing the playback from working. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream ATA 286 problems
Network-network-network I've sold and used them personally for about a year now. The Network is everything. Their soild on the right network or can be hell on the wrong ones. (some cisco systems). Also Asterisk is not a proxie or a switch. Some things don't work (like call forking.) I'm a sip provider and I like Sipura a lot better but the 3000 has problems with Asterisk. The VoIP Connection wrote: We have sold a lot of these adapters and we do have a few problems with them, but for every one that has problems there are at least hundred that work perfectly. Do we wish that they all worked perfectly? Of course. Luck of the draw I guess. Grandstream products have a one year warrantee. If you can show (with invoice or otherwise)that your product is in warrantee we will exchange it, regardless of where you bought it. All we will charge you for is shipping. Send an e-mail to [EMAIL PROTECTED] for RMA instructions. Michael Crown Managing Partner The VoIP Connection 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Andrejus Stavickis [mailto:[EMAIL PROTECTED]] Sent: Tuesday, April 26, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream ATA 286 problems Hi, Well, in my case I have a 486 and a hell lot of the problems with that. First I have not being able to use "features" like call transfer or anything like that (built-in ones) - it will just not respond to those commands. And Grandstream sent me a reply to my problem saying: "this is the problem with your * configuration and those features are not supposed to work without Asterisk". But eventually with the latest firmware I've managed to make it work after some voodoo. But the new firmware introduced another issue, which is kind of weird: my ADSL modem will resync every 10-15 min if I connect Grandstream ATA 486 to PSTN (do not get me wrong, I really put the ADSL filters in). As soon as I remove ATA 486, ADSL stays solid and does not resync. In your case you would just get a bounce backs from grandstream to your vendor and back, but in my case vendor will just not respond at all to any communication means. So beware VOIPSUPPLY.COM seems to be a bunch of funny people who will not stand behind the products they sell. So in my case I not just made a worst purchase, but also choose a worst supplier. Sincerely, --Andy x6722 I contacted the vendor I bought it from, and they said to contact Grandstream. I contacted Grandstream, and they told me to hit refresh in my browser After sending them the Ethereal trace, I haven't heard back from them yet. I think it's the worst purchase I've ever made. On 4/25/05, Anton Krall [EMAIL PROTECTED] wrote: Anobody had any problem with GS ata 286? The past few days Ive been having some problem with it, while making a call or during a call, I suddely hear a low noise like a car engine starting and then the ata dies, as if it got stuck or frozen. Anybody had these problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Audio sent using playback cmd
Hi All, I really need help on this. What would keep Asterisk from playing out audio files using the (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. In to Asterisk bun nothing coming out. Because I can hear the audio with the play tone I know there is something preventing the playback cmd from working. Thanks _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP, Asterisk and NAT
You are talking about a sip proxie server. I don't like ser. I use a full commercial proxie that works great but it's expensive. I believe asterisk can do what you want but I'm not sure. I use Sipquest for my services. I'm a provider. Irakli Natsvlishvili wrote: 100k question - does asterisk correctly handle following situations: 1. Asterisk is on a public IP Two SIP clients on separate networks, each of them are behind dynamic NAT gateway. Nat gateway does not have ALG. Media stream SHOULD NOT go thought asterisk. 2. Even worst case - three clients, two of them on one site, second is on another site. For example extensions 500 and 600 are on the same site and in the same subnet and extension 1000 is on another site/network. There are PAT FW/gateways with dynamic public IP in front of clients and those are symmetric NAT/FW. The task - clients registering on Asterisk server, calling each other and RTP should not go via asterisk. So, media stream should go directly from one client to another. I want to know: 1. Is it possible? - yes/no. Implementation should involve asterisk and SIP clients and not involving third party hardware products - ALG, session border controllers or so on. 2. If it is possible, what are requirements for SIP clients. 3. What configuration changes should be done on Asterisk server and on a sip clients. And final question - if it is NOT possible with Asterisk, do you know an open source product which works in above stated scenarios and you've actually tested it. Thanks for your help. Irakli ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Audio sent using playback cmd
=/O1uQjFNZZm0iZxLoUHo3USzsbg=,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=5c12ad4047730b1e8630f0c4df546f5f,qop=auth,nc=0002,cnonce=5756cc01 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 11 headers, 0 lines Sending to 208.41.254.119 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQAtDDRAijYtzRUbr0h3HNlvezg_;received=208.41.254.119;rport=5060 Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-44e76474 From: 6262769000 sip:[EMAIL PROTECTED];tag=413d98edc38224cfo0 To: sip:[EMAIL PROTECTED];tag=as350e5228 Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.41.254.119:5060 == Spawn extension (default, 9009, 2) exited non-zero on 'SIP/208.41.254.119-089b2aa8' Destroying call '[EMAIL PROTECTED]' Rod Bacon wrote: What errors are you seeing at the console? The only time I've ever had this problem was because I specified the file extension in the filename. Eg. Playback(file.wav) is INCORRECT. Needs to be specified as Playback(file). Some more info may help to get your question answered! - Original Message - From: Michael D Schelin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 10:55 AM Subject: [Asterisk-Users] No Audio sent using playback cmd Hi All, I really need help on this. What would keep Asterisk from playing out audio files using the (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. In to Asterisk bun nothing coming out. Because I can hear the audio with the play tone I know there is something preventing the playback cmd from working. Thanks _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why can't I hear audio?
Hi Everybody can someone tell me why I can hear audio? My call is to my proxie which is directing it to my Asterisk box. The Voice mail is playing but I think its playing to my proxie. the phone is on 198.31.185.246:63257 Here is from the sip debug. Thanks Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:63257;user=phone Supported: replaces Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, qop=auth, nc=0001, cnonce=1a605453cf8a557d, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=874d55e7960ad550b78bb1d8660faf69 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 338 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 asterisk1*CLI v=0 o=6262769011 8000 8001 IN IP4 198.31.185.246 s=SIP Call c=IN IP4 198.31.185.246 t=0 0 m=audio 63268 RTP/AVP 0 4 9 15 2 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:15 G728/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 16 headers, 15 lines Using latest request as basis request Sending to 208.41.254.119 : 5060 (non-NAT) Found no matching peer or user for '208.41.254.119:5060' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 198.31.185.246:63268 Found description format PCMU Found description format G723 Found description format G722 Found description format G728 Found description format G726-32 Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115 (g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 9009 in from-sip-external list_route: hop: sip:208.41.254.119;lr;hash=sipd-0-2-2 list_route: hop: sip:[EMAIL PROTECTED]:63257;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.41.254.119:5060 -- Executing VoiceMail(SIP/208.41.254.119-089aef50, 9009) in new stack We're at 208.41.254.125 port 13630 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 v=0 o=root 2330 2330 IN IP4 208.41.254.125 s=session c=IN IP4 208.41.254.125 t=0 0 m=audio 13630 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 208.41.254.119:5060 -- Playing 'vm-intro' (language 'en') asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQAOCCEtoOY6oOebox7ZBwoRRiY_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK2e698f7c1c332d46 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Contact: sip:[EMAIL PROTECTED]:63257;user=phone Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0002, cnonce=b85d4240018f156a, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=4030f97656e76c9bffecee6942efbfcc Call-ID: [EMAIL PROTECTED] CSeq: 55676 ACK User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow:
[Asterisk-Users] Playback dosen't play Playtone(Congestion) does ?
Hi All, What would keep Asterisk from playing out audio files (Playback command) but I can play the busy tone . playtone(Congestion) ?? I have verified this with ethereal and see the audio only going one way. Because I can hear the audio with the play tone I know there is something preventing the playback from working. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why can't I hear audio?
Hi Everybody can someone tell me why I can hear audio? My call is to my proxie which is directing it to my Asterisk box. The Voice mail is playing but I think its playing to my proxie. the phone is on 198.31.185.246:63257 Here is from the sip debug. Thanks Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:63257;user=phone Supported: replaces Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, qop=auth, nc=0001, cnonce=1a605453cf8a557d, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=874d55e7960ad550b78bb1d8660faf69 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 338 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 asterisk1*CLI v=0 o=6262769011 8000 8001 IN IP4 198.31.185.246 s=SIP Call c=IN IP4 198.31.185.246 t=0 0 m=audio 63268 RTP/AVP 0 4 9 15 2 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:15 G728/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 16 headers, 15 lines Using latest request as basis request Sending to 208.41.254.119 : 5060 (non-NAT) Found no matching peer or user for '208.41.254.119:5060' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 198.31.185.246:63268 Found description format PCMU Found description format G723 Found description format G722 Found description format G728 Found description format G726-32 Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115 (g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 9009 in from-sip-external list_route: hop: sip:208.41.254.119;lr;hash=sipd-0-2-2 list_route: hop: sip:[EMAIL PROTECTED]:63257;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.41.254.119:5060 -- Executing VoiceMail(SIP/208.41.254.119-089aef50, 9009) in new stack We're at 208.41.254.125 port 13630 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 v=0 o=root 2330 2330 IN IP4 208.41.254.125 s=session c=IN IP4 208.41.254.125 t=0 0 m=audio 13630 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 208.41.254.119:5060 -- Playing 'vm-intro' (language 'en') asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQAOCCEtoOY6oOebox7ZBwoRRiY_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK2e698f7c1c332d46 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Contact: sip:[EMAIL PROTECTED]:63257;user=phone Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0002, cnonce=b85d4240018f156a, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=4030f97656e76c9bffecee6942efbfcc Call-ID: [EMAIL PROTECTED] CSeq: 55676 ACK User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow:
Re: [Asterisk-Users] Re: TE110p - universal voltage?
Thanks all. I too have found out that the card is both. Mike Tony Mountifield wrote: In article [EMAIL PROTECTED], Craig Guy [EMAIL PROTECTED] wrote: Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and 5 volt pci slot? From photos it looks to be a universal card but the digium literature makes no mention of voltage requirements. I can cofirm that it has both the 5V and 3.3V cutouts in the edge connector. I can also confirm that I've used the card successfully in a 5V slot. I haven't tried it in the 3.3V slot. Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYOD provider other than broadvoice
Please give me a call 626-276-9009 I'm Mike Schelin of Shelltel a service provider in Southern Cal. Brian Capouch wrote: Wiley Siler wrote: Multiple providers... I am currently using one for outgoing exclusively due to the low latency and excellent call quality You mind saying who that is? Thx. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Hello Henry em=1-23 should be bchan=1-23 you have it set for analog also signaling=pri_cpe Henry Devito wrote: Don't you need one of these directives so the PRI knows which is master and which is slave? pri_cpe: PRI signaling, CPE side pri_net: PRI signaling, Network side Henry - Original Message - From: Scott Wolfe To: Asterisk-Users@lists.digium.com Sent: Friday, April 22, 2005 11:01 AM Subject: [Asterisk-Users] TE11OP - Mitel 200Sx?? Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk - TE110P -Kentrox (csu/dsu) - Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami em=1-23 dchan=24 Zapata.conf signalling=em_w switchtype=dms100 echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=default channel = 1-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$200 bounty for * paging feature
Ok you guys enough. The debate will go on forever. The only thing that seperates the boys from the men in this world is marketing. Beta vs VHS. Is Unix is better then Windows - Yes, but it doesn't matter. We live in a Windows world because Microsoft is the greatest marketing company on the planet! They also do somethng nobody else does. Tools Tools Tools. They made it easy to program on their system. If Unix and Lenux had the or can create the ease of use software tools, you will see the end of Microsoft. That is the key. Get the tools in the hands of all the programmers. A tool like VB on unix would just kill. trixter http://www.0xdecafbad.com wrote: On Wed, 2005-04-20 at 09:36 -0400, Walt Reed wrote: On Tue, Apr 19, 2005 at 06:24:09PM -0700, trixter http://www.0xdecafbad.com said: as a whole. I enjoy cheap computers, if it were not for microsoft creating windows, making computers easier to use for everyone, the mass production and highly competitive hardware market would not exist. If that didnt happen the $300 computer of today would likely not exist, and if it did it would cost more like computers did 20 years ago, $2000+ for a bare system. rantmode Um, that's total bullshit. Low computer prices and "ease of use" would have existed if MS was never around. You completely dismiss billions of man hours of hard work by those outside MS making advances in hardware and software around the world. To make a statement like that, you show a total lack of knowledge of the industry. and hoiw many operating systems were so popular during the 80s and early 90s? What operating system shipped on almost every computer during that period? I dont think I lack understanding of the industry I think that I remember clearly that windows was shipped on that, I think that whether or not it resulted in an anti-trust conviction microsoft did make it easier for people to use computers and thus more sold. I am sorry that you are so bigioted to think that other operating systems dominated the market during that period, and cant accept that windows was the #1 operating system by a clear margin in terms of installed systems. I have worked for over 10 years in the software development industry and Then you entered the industry far too late to know the real history of computing, have read too many MS revisionist history books, or were hiding under a rock. I started using computers in 1976. I dont think I entered too late. As for reading MS revisionist history books, no but I think that you have been readiung too many anti-MS revisionist history books. The popularity of a personal computer in the home was not made with cp/m it was not made with coherent (a unix for the pc before linux was around). It was not made by os/2, it was not made by any mac. Computers did not fully become so incredibly popular until windows. look at any historical sales reports and see when the numbers started increasing dramatically. Recall all the software shops that sold software, why was it that at least 90% was for windows and the remaining 10% for all other operating systems for a great many years? Why did all the computer shows that were oh so popular during that period sell mostly for the wintel platform? For example, The Amiga for example had a wonderful OS, great multi-tasking, awesome windowing interface etc. over 10 years before MS but it never sold as well. You fail to understand that its sales that drove the cost down. os/2 was better than windows at multitasking too, but again it didnt sell so well. Granted there was evilness by microsoft that resulted in antitrust convictions over some of that but you just proved how clueless you are. You know nothing if you try to bring up the amiga when we are talking about sales. And you try to say that I dont know what I am talking about? (some would argue longer.) Comodore didn't have a chance against the mighty combo of IBM, MS, Compaq. and other x86 hardware and software vendors in the business world (the Amiga was originally designed as a game machine and could never escape the stigma AND had the same bone-headed single hardware source issue that Apple has. Poor management / marketing also contributed to the companies death.) (Speaking of Apple, it boggles the mind that it took them over 15 years to add multi-tasking to their product line - and yes, I am dismissing their prior failed unix attempt.) You make excuses for the fact that they didnt sell as well as microsoft, and still try to insist that I dont know what I am talking about when I say that MS sold more units which drove the cost down (I specifically made that point in my previous email). MS has no effective competition due to their illegal business practices, killing off alternatives (BeOS is a recent example) by pressuring large ISV's to only write for the
[Asterisk-Users] TE110P card installation errors
ux-2.4/include/linux/module.h:192: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/module.h:193: `inter_module_get_request_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/module.h:193: warning: parameter names (without types) in function declaration /usr/src/linux-2.4/include/linux/module.h:194: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/module.h:194: syntax error before numeric constant /usr/src/linux-2.4/include/linux/module.h:194: `inter_module_put_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/module.h:194: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/module.h:203: `try_inc_mod_count_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/module.h:203: warning: parameter names (without types) in function declaration In file included from /usr/src/linux-2.4/include/linux/fs.h:19, from /usr/src/linux-2.4/include/linux/capability.h:17, from /usr/src/linux-2.4/include/linux/binfmts.h:4, from /usr/src/linux-2.4/include/linux/sched.h:10, from /usr/src/linux-2.4/include/linux/mm.h:22, from /usr/src/linux-2.4/include/linux/slab.h:14, from /usr/src/linux-2.4/include/asm/pci.h:38, from /usr/src/linux-2.4/include/linux/pci.h:671, from tor2.c:33: /usr/src/linux-2.4/include/linux/dcache.h: In function `dget': /usr/src/linux-2.4/include/linux/dcache.h:254: warning: implicit declaration of function `__out_of_line_bug_R8b0fd3c5' tor2.c: In function `tor2_spanconfig': tor2.c:209: warning: implicit declaration of function `printk_R1b7d4074' tor2.c: In function `init_spans': tor2.c:277: warning: implicit declaration of function `sprintf_R1d26aa98' make: *** [tor2.o] Error 1 Can someone tell me what is going on? Thanks Michael D. Schelin SHELCOMM/ ShellTel 626-814-2354 or 626-276-9009 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P
Ok I [EMAIL PROTECTED] up. I didn't realize the card is 3.3 volts and my new computer is 5V. Can anyone point me to a PCI to PCI bridge. Any suggestions? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P card installation errors
I discovered my computer is 5v and the TE110P is 3.3V Could these errors be because there was no card? Michael D Schelin wrote: Hi All, I just installed a TE110P card and I'm trying to compile the code. I followed to the letter the instructions. This is what happens. [EMAIL PROTECTED] zaptel]# make clean rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f *.ko *.mod.c .*o.cmd rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c zaptel.c cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:61: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:62: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:62: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:63: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:63: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:71: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:71: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:71: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:71: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:73: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:73: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:75: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:75: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:76: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:76: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:78: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:78: syntax error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:78: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:78: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:80: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:80: syntax error before numeric con
Re: [Asterisk-Users] Any work around for ISPs that block port 5060 and69
I'm a service provider and my system does not requior 5060. See if your provider can use other ports. I would think asterisk server can be remapped. They can't block them all! Kanuri, Seshu (Company IT) wrote: Yes, there is a solution. Use IAX2 both on Server and Clients and bypass all that Port mapping crap. There are a few brands of VOIP Phones available in the market that can do IAX2. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Jn-Francois Sent: Tuesday, April 19, 2005 9:48 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Any work around for ISPs that block port 5060 and69 I have a several friends registered on my asterisk box that are experience problems with their ISP blocking SIP default ports 5060 and tftp port 69. Is there any way around this problem or are they forever doomed to VOIP since their ISP is pretty much the only ISP company on that island. So far I was able to have them change their default SIP port to 6070 and any packets coming in on that port on my asterisk box I would redirect to port 5060. That seem to be working fine, expect that they can make calls but cannot receive calls. I think part of the problem might be that when asterisk tries to initiate a call to their sip phone it tries on port 5060 instead of 6070 even though I have specified in the sip.conf that their port is 6070. Has anyone else encountered this problem and was able to resolve it? My other option is to change my asterisk box to work completely on a different port, but I am reluctant to do so since the majority of registered users for now do not have an issue with port 5060 being blocked by their ISP. Thanks for your help. Joel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] large analog to asterisk
Good point. Here is another Suggestion. Why not use the existing analog phones to their PBX and go out to channel banks for their phone line trunks. Then go to Asterisk for the rest. They don't have 700 trunks. This will save on equipment costs and you will get some of the benefits of Asterisk. It's not the best solution but it will work. Also if you do want to replace all you system with new phones then try my idea of using the cat 3 cable. As far as the switches gos, remember it's not the cable the determines the speed but the equipment connected to it. I have yet to see a sip phone above 10 Mb. So you can disregard Mr. Hamilton's statement about the switch. Yes you investment will be high, but that is a business expense. I'm currently doing that cat 3 trick. Don't worry about your customers connecting to your phone system. They won't know it's IP. John Novack wrote: Andy Hamilton wrote: And then you'd need to purchase 700 VoIP phones; not a small investment. With all due respect to Mr. Schelin, I think the analog method may be best, unless you plan to expand the services that you offer to the guests. If the rooms did have cat3, you could eventually expand your offering to include internet access for the guests, advanced phone features (on the IP phones), etc... Even Cat 3 anymore could be a real problem Most inexpensive hubs and switches are 10/100, with no way to lock them to 10 . Wear and tear on SIP phones, most hotels are used to paying 12-15 bucks for room phones, and some always end up walking away.. And then there is the issue of E911. Depending on the location, some jurisdictions, and I feel sure many more to come, are requiring E911 to know not only the street address but the room number. Good luck. Stick with the analog for now. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] large analog to asterisk
Oh one more thing. There is a 300 foot limit to Ethernet. Also the minimum number of wires is 4. shane fowler wrote: we are looking at the ability of being able to convert large phone system over to asterisk or if it's possible at all. The building is two sections containing a large office section (with data cabling) and the second section is a hotel with no data cabling. The first section is a no brainer with sip hard and soft phones but the hotel part is where the problem lies. The current count of rooms in the hotel is about 600...that's at a minimum 600 analog connections. Some rooms have 2-3 phones so as a rough number i'm saying 700 total. I see where some people use the Adit 600 to do up to 48 analog connections that trunks over 2 T1 connections back to asterisk but for 700 phones thats 15 Adits with 30 T1'show in the world would you do that?? just several asterisk servers with 2-3 Adits per server? is there any other way? I'm open to suggestions. Thanks.. Shane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] large analog to asterisk
If your rooms analog phones are wired with cat 3 cabling you can do 10 Mb over it. Convert all the rooms to Ethernet and use large switches. One Asterisk box should do the trick. Remember not every room will be using the phone system at the same time. This should work for you. shane fowler wrote: we are looking at the ability of being able to convert large phone system over to asterisk or if it's possible at all. The building is two sections containing a large office section (with data cabling) and the second section is a hotel with no data cabling. The first section is a no brainer with sip hard and soft phones but the hotel part is where the problem lies. The current count of rooms in the hotel is about 600...that's at a minimum 600 analog connections. Some rooms have 2-3 phones so as a rough number i'm saying 700 total. I see where some people use the Adit 600 to do up to 48 analog connections that trunks over 2 T1 connections back to asterisk but for 700 phones thats 15 Adits with 30 T1'show in the world would you do that?? just several asterisk servers with 2-3 Adits per server? is there any other way? I'm open to suggestions. Thanks.. Shane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID reseller structures
If your in Los Angeles Call me I've got 130,000 numbers with caller ID from my ss7 network. Trust me there's a whole lot more to it than what he just said. Mike trixter http://www.0xdecafbad.com wrote: On Thu, 2005-04-14 at 19:37 -0700, snacktime wrote: I knew xo/level3 were clecs, and that the numbers came from nanpa, but I didn't know the requirements for getting numbers. So theoretically anyone with some type of switch can go to nanpa, get a CIC and some numbers, and then get someone like global crossing to terminate everything to you? maybe once the FCC ruling goes a little further, right now its limited to sbc only. Basically to get numbers you have to have a OCN and a CLLI code. If you are going to interconnect with SS7 you need a point code. (required for number pooling, where you get only 1000 numbers instead of 10,000, which is the prefered way to do it). To get that under the current regime you basically have to be a LEC/IXC. NANPA wants a minimum of 66 days to assign numbers and you have to have them entered in BRRDS which they will do for $35 first year less next year (there is no charge for exchanges from NANPA but resellers may charge you since its not a trivial undertaking to get numbers, and if you wanted all 50 states you would have to be approved in all 50 states by each state PUC/PSC/BPU whatever they happen to call it). Hope this helps :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 503 Service Unavailable
What would cause this error. My server is not busy. I'm trying to ger the voice mail to work without any PSTN extensions or cards. Just a sip Mailbox. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice mail playback
Hi all How do i set up voice mail playback using * as the inertupt . I can't seem to figure out hou to use VoiceMailMain([EMAIL PROTECTED]) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice mail playback
Hi all How do i set up voice mail playback using * as the inertupt . I can't seem to figure out hou to use VoiceMailMain([EMAIL PROTECTED]) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk management portal
Hi everyone, Why doesn't this work? I can't get in. Is it because I changed the root? User: admin Pass: password ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel P2000W Finally (Almost) Working
Your right on the overpriced junk! But yes now it works great. Rod Bacon wrote: I found new firmware at: ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip The phone is now finally (almost) useful. Still a cheap piece of crap, with new bugs to replace the old, but at least it sort of works now. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring the Sipura for static IP and registering with Asterisk.
There is very little difference between configuring a static IP or DHCP. You need the basic 3 things like the IP address, Sub net mask, and Gateway address. For DNS use the dns servers address's supplied by you ISP. Make sure you turn on the use DNS setting in the Sipura unless you use IP address only. A sip registration failure can be many things. Your Service provider should have given you basic settings. Any one of them can be typed in wrong. You must go over each setting. A common failure is the auth. password. Also make sure you use a stun server. Even if the IP is public, It doesn't hurt anything to use it. Most of the time stun servers use port 3478. So your entry should be ip-address:3478 I hope this helps a little Mike Jerry wrote: OK so now you have an IP address. Did you login and configure the Sipura? On Apr 7, 2005, at 1:04 AM, Rich Adamson wrote: I wish to configure my Sipura with static IP. I have set the static IP, but there is registration failure on doing so. Could you please tell me how do I go about configuring my Sipura for static IP and register it successfully with the Asterisk server. A few of the spa changes require the box be rebooted. Did you do that? Can you ping the sipura's ip address? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA and NAT traversal
The only way you ll be able to call extension to extension is if Asterisk is on the same node behind the nat. like the extensions or if each extension is on a different node. I run a proxie server and have ran through this problem many time. I bet you can call out bound to the outside world just fine from every extension. . Eric Wieling wrote: Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax with SIP devices
I'm a Voip carrier and we support T38 calls. If you need T38 Faxing Please contact me [EMAIL PROTECTED] Mike Steve Underwood wrote: Asterisk needs more than to just allow T.38 as a codec type. T.38 only recently gained an RTP option, and very few things support it so far. You need to have a UDPTL transport for most boxes supporting T.38. I have a working UDPTL for *, but it needs more polishing before release. Regards, Steve Mark Dutton wrote: Right you are Michael. I have some Multitech MVP200s and they do work indeed. Only problem is mine are too old to do SIP. I know Asterisk does not do T.39 but as it only needs to ALLOW the codec when devices are communicating with each other, it can't be too hard to get working. Perhaps the t39fax codec needs to be added to the Asterisk codec list so it knows about it and then it can be added to the allow list in SIP. Mark Date: Thu, 07 Apr 2005 21:17:03 -0700 From: Michael D Schelin [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T.38 fax with SIP devices To: Scott Wolfe [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello, The Multitech VOIP line supports T38 and I have tested it. It works great. You will need a public IP to make it work. Very expensive though. T38 Is not compatible with Asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Beeps during Sip to Sip phone calls
Grandstream has the same problem. Very common. A simple DTMF debouncer curcuit will fix it. Doug Meredith wrote: Eric Wieling [EMAIL PROTECTED] wrote: Daryll Strauss wrote: Yep, I've seen it and from reading http://www.voxilla.com it's a pretty common problem. If you turn on debugging what you'll see is that the Sipura has mistakenly detected a DTMF code in the audio stream and is relaying it by repeating the signal (very loudly I might add) So this appears to be a bug in the most current firmware. I've reported it to Sipura including the debug output. Maybe more people should do the same. You'd think that switching to RFC2833 DTMF would fix that. That is actually the problem. It thinks it hears DTMF so it sends an out-of band signal. The other end receives this and produces the audible tone. Switching to in-band fixes it. Well, works around it. :) Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting a good deal on a PRI
I may be able to help. I'm a provider in Southern CA. What you need to do is eliminate all pots lines by moving them over to VOIP completely. This will take some time but will save you a lot of money. Please call me for more info. I can provide you service and if your interested LNP your numbers. I would have to check. My number is 626-276-9009. If 90% of your calls are inbound you should not be paying much at all. Michael Welter wrote: snacktime wrote: We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The price they quote is not the end of the story. I just received ny first invoice from my CLEC for a voice T-1, and the taxes and other charges are significant. Be sure to ask what the total amount will be. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stand alone Voice Mail
Hello everyone, I need to configure a stand alone Voice mail box. Calls will come in via sip. I have read and read until my eyes hurt for 2 weeks now. Can someone email me the basic config files needed to do this. The examples are overly complicated. I just need a simple basic configurations without all the clutter. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting a good deal on a PRI
If you feel your pots lines are critical to your business then you should keep them. What I want is the flexibility and features * gives me, with the reliability of land lines Then why do you need a PRI? You can use a channel bank to convert to PRI if you have DID's. There is a lot you can do to get calls into the asterisk box. But if you feel you need the best reliability why use asterisk. I don't feel it's 100% yet either. It's good, but now you have another source of possible failure. No more or no less than VOIP. I'm a service provider and yes we have failures once an awhile but I can tell you in about a year or so that will be a thing of the past. snacktime wrote: On Apr 7, 2005 6:32 PM, Michael D Schelin [EMAIL PROTECTED] wrote: I may be able to help. I'm a provider in Southern CA. What you need to do is eliminate all pots lines by moving them over to VOIP completely. This will take some time but will save you a lot of money. Please call me for more info. I can provide you service and if your interested LNP your numbers. I would have to check. My number is 626-276-9009. If 90% of your calls are inbound you should not be paying much at all. voip just isn't reliable enough yet to get rid of land lines completely. I feel comfortable using voip for outgoing calls, and maybe 800 numbers we use for marketing campaigns, but we can't afford to have our customer service lines down. Plus, another issue is adequate bandwidth. If we go pure voip I doubt our dsl would be enough (1792 down/448 up). Which means I'd have to get a T1 or better for the office. What I want is the flexibility and features * gives me, with the reliability of land lines, and also the ability to interject voip into the mix if and when it makes sense. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
