Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
On 05/11/2011 08:39 AM, asterisk-users-requ...@lists.digium.com wrote: Snore... Now move along please... OK, but how about you respecting some basic mailing list etiquette and not quoting the entire thread in your posts so that those of us who wade through these messages as digests don't get carpal tunnel with the scroll wheel on the mouse sifting through your pissing match with another on this list. Take it outside. Your welcome. Myles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, SIP Firewalls
Hi all, I'm trying to get my head around our Asterisk network configuration. We've been using it for about 2 years now (home office) and it works great. Its Asterisk 1.4.2 with SIP through external provider(s). We have the Asterisk server behind our IPCop firewall, and have a dedicated IP address that comes to the firewall from our ISP (Cox) and that is routed to our Asterisk box using SIP ports, etc. It works fine, connects without issue and we then have all of our SIP Phones throughout the house for the calls. My wife I run businesses from our home, so we have multiple numbers coming into Asterisk and with some fancy Asterisk scripting, etc. we have the one system acting as a phone system for 4 companies. Works great. Well there is one 'optimization' that I need to sort out. There seems to be some latency between the Asterisk server (and the SIP Phones) and callers. Depending on the caller's network (ie. POTS, Cell phone, other Voip, etc.) we find about 30% of the time that there is a small delay (about 1/2 a second) between us talking and the caller hearing it, which makes it sound like the caller is talking to an offshore company located in South Asia. I have read numerous posts, discussions, etc. about this sort of thing and it seems that it has something to do with our Firewall, QoS, etc. and I'm entertaining moving the entire Asterisk server outside of our Firewall, and connecting the SIP phones to it on an entirely separate sub-net with a dedicated NAT router. It kinda scares me though. I know that SIP is an attractive attack-vector, and that there are scripts out there that target SIP devices. I know I could run Fail2Ban on the server, which is fine (we're doing that anyway now), but before I go down this path, I wanted to get general feedback if we are using our Asterisk system using 'best practices' or whether it should never be sitting behind a Firewall, despite the fact that it is working pretty close to perfect as it is right now. I just want to find a way to reduce the latency. Does anyone have any thoughts about this? Thanks in advance for any comments or suggestions. Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA LLC www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird phone behavior after recent CentOS 5 update
For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to voice mail. But there is no audible 'ring' to the caller. Just nothing - blank, empty silence. Of course any automated answering system (ie. business phone menu, etc.) that we have works just fine. Its just the lines that go directly to an internal phone that are no longer providing any audible ring which is sending a message to the caller that their call didn't go through. Does anyone have any idea what might cause this? Myles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird phone behavior after recent CentOS 5 update
Some more info on this weirdness Seems that if I play any audio out from Asterisk first the problem goes away. its almost like the entire audio engine isn't being 'initialized' or something on those direct calls. I found that if I execute a Swift(Please Wait) before I call the Dial command in my extensions, then it plays the ringing tone as expected. But if I receive a call, and then try and have it transferred directly to a phone extension via a Dial command (which used to work perfectly well), the calling party never hears a ringing tone. But if I play some audio purposely before, then it rings Myles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exporting Blacklist database
Is there a simple command in the CLI or other for Asterisk 1.4.2 where I can list all numbers in the blacklist database? I need to export this data to another database, but am unsure how to get to it all in a list. Thanks in advance for any pointers. Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Asterisk/SIP call forwarding behavior
I have a small Asterisk 1.4.2 system that I run out of my home based business, and my Dialplan has it set to send any incoming call to my desk Grandstream phone. Works really well. When I leave the office, I need to re-direct the calls to my cell phone. I tried to do this through my Grandstream phone, but it really didn't work all that well. Callers would tell me that they tried to call but got cut off, etc. So I decided to setup my dialplan with a variable in it for the destination phone to route to (SIP/MyDeskPhone or Local/15) (number excluded for obvious readons) to my cell phone. What I'm finding, however, is that my Dialplan has about a 50% chance of successfully routing the phone call through to my cell phone when I'm switched over to using it. I suspect it has something to do with a timing issue with my outgoing SIP provider or something like that. What I'm looking for is some sort of advice on 'best practice' to handle call routing to my cell phone vs. keeping it on my LAN to my desktop phone. As I mentioned, the desktop phone works flawlessly and I'm trying to get the same results on my cell phone so I don't lose any calls in the process. Thanks in advance for any advice. Myles -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HElP me I am a beginner
Prashant wrote: I am completely new to the asterisk so can any one help me with it as I have some questions queries 1. first n for most what are the tools/equipment that I need for eg a Computer, and a net connection is it all that I need for simple head start to get hands on the asterisk Hi Prashant, When I started working with Asterisk, I was impressed with the wealth of information out there on this area, but also overwhelmed by the amount of it and probably spent 50% of my time trying to reconcile one piece of information I found that was 5 yrs old with another piece of information I found that was 2 yrs old. The sheer amount of information that you will find means that you won't be caught out with no answers. But trying to make sense of which answer is the best fit for you is the challenge. As the other posters have rightly stated, getting a comprehensive single book as a starting point is the best place. The author would have collected and documented their information in one set that all works together. Although the general community favor 'Asterisk: The future of telephony' which is an excellent book, there are multiple editions of it so you want to make sure you get the latest one. I found this book to be my 'savior' when learning Asterisk: Practical Asterisk 1.4 1.6: From Beginner to Expert by Stefan Wintermeyer The reason why this has become my #1 favorite is simply that it explained a lot of telephony concepts as they apply to Asterisk - often these are assumed with setting up a VoIP solution. I think the book can be found here: http://my.safaribooksonline.com/9780321670632 There is an online version and a print version you can order. I understand that spending money on this sort of thing in some economies is a major expense, however the time saving you will get trying to learn this product is far worth the investment in the book. Hope this helps and good luck! Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Checking blank CallerID in Dialplan
I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: anonymous So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was implement some logic that checks for actual numbers in the caller ID. We have a couple of different SIP providers for incoming calls. Some prefix numbers with a + and others don't. But I'm logging incoming calls that are getting through our tele-spam filter and it seems that they are blank, but I suspect they contain empty spaces which is why our matches don't work. Does anyone have some sample DialPlan code that they are using to thwart incoming calls with no caller ID? I was thinking of maybe converting the caller ID num to a numeric value and testing for 'not equal to 0' but that won't work with the + prefix. All suggestions greatly appreciated. Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manipulating the Blacklist database
I am running Asterisk 1.4.2 and recently we changed the SIP provider of our main incoming DID number. The new provider prefixes all CallerID records with a +1 in front of the number, whereas the previous SIP provider did not. Consequently now all my blacklisted numbers aren't matching in my Dialplan, so I'm getting tele-spammed. Is there a way that I can work with the blacklist database like a SQL database, and just apply a script to update all numbers and add the prefix to them if they don't have it already? Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Working with Blacklist database
I have an Asterisk 1.4.2 system online and have built up quite a large blacklist of tele-spammers that have been calling us. Recently we swapped one of our DID numbers to a SIP provider that now prefixes all calls with +1 in front of US numbers (we're in the USA) and I need to edit my blacklist database so that all numbers in there that don't have the +1 prefix in front of them, now have a version of it that can match to these new prefixes. Or alternatively I need to change my Dialplan to search for a match against the Blacklist database on a substring of the incoming caller ID. Has anyone done this before? Is it easy to edit the content of the Blacklist database like in a SQL database, where I can simply insert new records based on the old records but change the prefix to them with the +1? Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cell phone redialer?
I have an Asterisk 1.4.2 system installed at our office, and have a few 'on the road' sales people that want to make calls from their cell phones in transit, but they are complaining that people returning calls that they make from their cell phones are simply just using the CID that is coming from the cell phone which is causing them to get phone calls outside of business hours. What I was hoping was that either there was an existing Add-On to Asterisk out there for this, or that I could construct something. Basically I wanted to see if I could get them to call our phone number on Asterisk, enter some special extension and/or enter a passcode, and then enter the phone number that they wanted to call in which our phone system would route the call to the party but with our caller ID. Is something like this possible? I know there are companies out there charging for this sort of thing (like SpoofID, etc.) but since we may already have the technology in place for this, I was hoping I could roll my own 'home grown' version of it. Thanks in advance for any comments. Myles - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplans Holiday Dates
I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far. Does anyone have a suggestion as to how to approach this? I'm running Asterisk 1.4.2. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring SIP Skype connections
I have an Asterisk 1.4.2 server with 3 different SIP providers and Asterisk for Skype gateway installed. Periodically the SIP providers go offline for some reason, or the Skype connection fails. When this happens, I lose my SIP registration to the provider. Unfortunately I don't know this has happened until someone eventually contacts me to say, I tried to call you but it didn't work. This, of course, sends the wrong message to the caller and I have no way to determine if I'm using a reliable SIP provider or not as I have no statistical log of their uptime. What I'm trying to find is how others monitor the uptime status of any external SIP providers or gateways that they use. Are there tools or add-ons available for this that will email me when a SIP registration goes offline? Any suggestions for this would be greatly appreciated. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for some example dialplans
I have an Asterisk system setup for our small business, and its working well. I posted to the list about a week or so ago, regarding having it handle direct extension dialing, and unfortunately I'm not any closer to solving this issue, so I was hoping someone might have a working example of how to set this up they could point me towards. Basically I have everything EXCEPT direct extension dialing working. What I mean by this is that a caller gets an ACD menu when they come into the phone system, (1 for sales, 2 for support, 3 for customer service, and 8 for directory of employees). I have setup extensions with 4 digit numbers, all beginning with 6 (ie. 6001, 6002, etc.). I was hoping that I could capture the first digit entered as a 6 which doesn't correspond with a menu option, and then get the next 3 digits and dial the appropriate extension. Unfortunately that's not working because the system isn't quick enough to process and deal with additional digit entry without dropping a digit or two. I was told to use wildcard maskings like this _ for the extension, which I attempted over the weekend. Unfortunately the second I hit a digit that wasn't in the menu options, it dropped the call. What I have that is working at the moment is this: ; Calls during business hours exten = s,1,Set(TIMEOUT(digit)=1) exten = s,n,Wait(1) exten = s,n,Background(01-welcome_mod) exten = s,n,WaitExten(3) exten = s,n,Goto(ts_operator,s,1) ; User pressed 1 - sales exten = 1,1,Wait(1) exten = 1,n,Goto(ts_sales,s,1) etc What I need to know is what to change this to (exactly) so that it will allow up to 4 digits to be entered, but on the first one being entered, it is validated against the phone menu options. Does anyone have any example dialplans that show how to do an ACD menu, with direct extension dialing supported as well they could point me towards? Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help/suggestions for DialPlan
Steve wrote:
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,goto(support,s,1)
exten = 3,3,goto(customer-service,s,1)
exten = 8,3,goto(directory,s,1)
exten = _x,3, hangup()
exten = _6xx,3, goto(dial-by-extension,s,1)
First, thanks to everyone for the pointers.
This looks like it might fit my ACD requirements. What I'm still
confused about here is that it seems to get the extension in line 1, but
answers in line 2? Wouldn't it have to answer the call before it could
get an extension?
And what I'm also not sure about is how this would work based on the
number of digits entered. Is it that by not specifying how many digits
to get, it just handles processing 3 digits entered if the first digit
is not 1,2,3 or 8?
Myles
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[asterisk-users] Need help/suggestions for DialPlan
I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime to directly dial that extension. I am setting up extensions to be 3 digits long, all starting with 6 (ie. 601, 602, 603, etc.) My current setup requires a single digit to be pressed before going to that context in the Dialplan. Its all working fine with the exception of direct employee extension dialing. What I have done is to create a context for '6'. In other words, I wait for a single keypress at the main menu and if I get a 6, I then transfer to get the next 2 digits which represents the employee ID and then transfer to that person. But what is happening is that the speed in which Asterisk is picking up the digits from the caller isn't fast enough for some reason, and is often missing the first digit (6) and then processing on some other digit that it picks up on. I don't want to have to change our recorded audio menu since that was done professionally and was expensive, but just want it so that the incoming keys pressed are processed instantly. I'm using a TDM400 from an analog line for the incoming menu, and then transferring to our internal LAN once the extension is chosen via SIP. This is the context of my 'main menu': ; Calls during business hours exten = s,1,Set(TIMEOUT(digit)=1) exten = s,n,Wait(1) exten = s,n,Background(1-MainMenu) exten = s,n,WaitExten(3) exten = s,n,Goto(ts_operator,s,1) I have the context for extensions handled like this: ; User pressed 6 - start of extensions [ts_extensions] exten = s,1,Set(TIMEOUT(digit)=2) exten = s,n,WaitExten(10) exten = s,n,Hangup() exten = t,1,Hangup() exten = i,1,Goto(ts_start,s,1) exten = 01,1,VoiceMail(6...@edgeneering) exten = 01,n,Hangup() exten = 02,1,VoiceMail(6...@edgeneering) exten = 02,n,Hangup() exten = 03,1,VoiceMail(6...@edgeneering) exten = 03,n,Hangup() (I'm dummying out the transfer to SIP phone parts with VoiceMails here for demo purposes) What am I doing wrong that is stopping correct identification of the digits entered by the caller? Thanks in advance for any assistance. Myles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. We're running IPCop (Linux based, open source, 100% free), and its been fantastic for us. www.ipcop.org I spent weeks trialing many others. Even had Astaro send me out a trial box to use. I think we short-listed this down to pfSense, SmoothWall, Astaro and IPCop. Its been a while since we did this, so newer versions might have different test results now, but (if I remember correctly): 1. pfSense - Solid, but was a bit picky on network adapters (we wanted to use a Quad NIC for this). Also was a bit cryptic for setup, but that's probably just us being too lazy to RTFM. 2. Shorewall - this worked out of the box, looked easy to setup, etc. But when it came down to supporting multiple external WAN IP addresses that we had, it fell short and was dismissed as an option. I believe that their commercial version did support this, but had a hard time trying to find who to buy the damn thing from. 3. Astaro - great company to work with. Really helpful, great tech support, etc. Loving all of that. Not loving the $2K+ price tag for what we needed. But then we are stingy and cheap. That's just us. If you have commercial clients, and budget this looked really good. 4. IPCop - its free. Was a dream to install and setup. Support via their mailing list was awesome. The people there didn't make us feel like newbs when we had basic questions to ask. Feature set rivaled all other products, and there is a pretty healthy add-on market for it. QoS was decent, although there are add-ons for better QoS granularity. We chose IPCop. Been running it with Asterisk and our other network apps, servers, etc. for about 4 months straight. Never needed a reboot. Never crashed. Low footprint, and runs on some old dog hardware we had lying around. Like I said, this review is about 6 months old, so things change. That's our biz. Go figure. Of course, your mileage may vary. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What are the reasons for VoIP echo?
I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What are the reasons for VoIP echo?
Ira writes: Very similar to what I have. Also Flowroute for outgoing but others and a TDM400 for incoming. Since upgrading to 1.6.2 from 1.2.28 or so and figuring out DAAHDI and HPEC on the new version there have been no echo issues at all. Also cable modem but only the slow version. There is a Linksys router between the Asterisk box and the cable modem. Usually VOIP echo is the other end. There should be no way other than having the volume on your SIP phones way too high to get local echo. In the past it's been suggested on inexpensive phones with this problem to take apart the handset and make sure the path from the speaker to microphone is blocked or filled with sound insulation. I'm using Aastra 480i phones and that's never been a problem. That's interesting. Our setup is almost identical to yours. We have 3 different SIP phones, one being an Aastra 480i. I have a Grandstream 4 Line one, and my wife has a Grandstream 2 line one. Both are relatively high-end phones (not inexpensive ones). The echo issues we've seen don't appear to be any different regardless of the phones in use. And strangely we never actually experience them on our end. The problem is always reported to us by someone calling in to us. This is what suggested to me to migrate my DID numbers over to them, rather than keeping them with Broadvoice. I think I might do that as a precaution anyway since its not likely to be a big problem for us to do it. The volume on our phones are not loud. I use a headset with my phone most of the time, which seems to give a better quality call to the other end, but there's not an issue with volume that could cause it. I am curious about the fact that you said after upgrading to 1.6.2, your problems went away. I didn't start with that version because it wasn't the current production version at the time. Do you think it would be beneficial to migrate to that version for me? Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for recommendations - US SIP provider for T.38 Faxing
We recently completed a full migration of our office phone system over to Asterisk and are using flowroute Broadvoice as SIP providers for our incoming/outgoing calls. Everything is working great. Now that the phones are done, I'm left with the fax machine. Although we have been using jFax for digital fax handling for most of our fax needs, it falls short with many calls where the recipient fax machine takes ages to answer a call, or international fax machines aren't well supported. I'm ready to move this over to Asterisk/SIP as well, and I know that Asterisk has a Fax solution available. I'm just looking to find a SIP provider that gives really good results with T.38 fax handling (mainly outgoing fax calls). Does anyone who has done this in the US have a recommendation based on their own personal experience? Thanks in advance for any info. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc insert with mssql
David wrote: When I do an odbc show, it shows that I am connected to the db. If I use isql, I can write to the db, however, when I use func_odbc, a record will not write. I'm using asterisk 1.4.9. Any idea what might be wrong? Sounds sneakingly like a permissions problem in your database for the login account that you are using through ODBC. But if not, the INSERT SQL statement may not be correctly established for your database. Personally I don't use ODBC for this sort of thing. Its not really a 'Linux' type thing, and relies on a lot of middle-ware layers between the Asterisk server and the database. Not that you have too many choices, but what we do that works brilliantly well here is to use PHP-AGI to construct a PHP script that does all the DB connection, inserting, etc. and just have the Asterisk dialplan call that. This way I have total visibility to the SQL and I have apply some transforms to the data in PHP if needed. It also means that if I want to swap out the DB to another brand, I can do that with only one place to change the code. FWIW... Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc insert with mssql
Tilghman Lesher wrote: You've just stated the primary reason for folks to use ODBC. Perhaps you've written off the technology too soon and for the wrong reasons. You maybe right as I've been a SQL developer for about 25 years (actually before ODBC was ever around) and I might be tainted by spending countless hours of pain suffering dealing with it on a variety of non-Microsoft platforms. Off memory, and this is from a LONG time ago, I believe ODBC was originally created by Microsoft SIMBRA back in the early 1990s, hence my comments about it not being a 'Linux type' technology. In its early incarnation it had a reputation of being bloated and lacked in performance. I'm sure that 15 yrs or so later they must have gotten that under control. My biggest negative to ODBC is that the calls that the client application has to make to the database are hard to visualize and debug, particularly if you are crafting a SQL QUERY statement that goes on for pages. Its for this reason that I'd want to have something where I could get more visibility to the DML. It sounds like this might be the exact problem that the original poster was having - that their DML had errors in it, and if it could be visualized before it hit the database for debugging, it might have been quicker to debug. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set PHP binary location for AGI
I am not finding anything relating to this on Google, so I thought I'd pose the question here... I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a custom built PHP5.2.10 install to interconnect with our Firebird SQL database, which I've done. But I noticed that the default install path for PHP5 on this box appears to be /usr/local/bin/php rather than the path that the default PHP5.16 path of /usr/bin/php. To be certain that the correct PHP binaries are being called, is there a conf setting somewhere that I can tell Asterisk AGI where the PHP binary is that I want to use for this? I noticed that most AGI PHP scripts begin with: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q for my build, but is this enough? Is there somewhere else that needs to be updated in order for AGI to correctly find the PHP build I've done? Myles P.S. I did try and change the PHP configure options to install in /usr/bin but that didn't work because it won't install the man pages and headers in there for some reason. But its installing fine in /usr/local/bin which appears to be its default install location anyway, so I'd prefer to go with that and keep this simple if I can. -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring Asterisk uptime
We have added Asterisk to a line of 'mission critical' servers at our business, and being in the web application development business one of the core things we do is to monitor web server availability. I'd like to add Asterisk to the servers that our monitoring systems are handling, and also that our SIP trunk provider has our Asterisk system correctly registered at all times. What are the 'best practice' tricks used for monitoring an Asterisk phone system for uptime and SIP registration from an external monitoring server? I can certainly ping the box, but I really need more than this. I need to know if the Asterisk service is running, and also that there hasn't been any issues with SIP registration to our external trunks. If anyone could share how they are doing this sort of thing, it would be greatly appreciated. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring Asterisk uptime
David writes:
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep
USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host
keys to do this without user interation), you should be able to poll
any linux server for really anything you want.
Someone who is in the business of selling hosted applications should
be able to EASILY use awk and grep to figure out if his sip trunks are
in the 'Registered' state via SSH... Unless I misunderstood the nature
of your question and you were looking for something native to asterisk
or the AMI.
Thanks for the feedback, and its a good idea! We are monitoring about
45 servers right now with some 3rd party monitoring software, and
looking to move it all to Nagios. Unfortunately our currently software
doesn't give me as much flexibility on a per device basis to have it
execute a script for each device that its monitoring to this level, but
I'm thinking that we could customize Nagios to do this sort of thing.
Its definitely worth exploring and I'll post back with the results.
Thanks again.
Myles
--
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
http://www.techsolusa.com
Phone +1-480-451-7440
___
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Re: [asterisk-users] Set PHP binary location for AGI
Steve writes: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q This should work. Did you try it? Yes, its working fine. The only problem is that when I went to 'uninstall' the standard PHP installation that came with CentOS 5 on this box, it de-registered PHP from executing correctly but left all the old files still in place. So when I re-installed a new PHP build, I can't see what version is actually running from AGI. I'll get AGI to do a phpinfo() call and see if I can pipe the results to a file so I can see what is going on here. From what I can tell, however, it would appear that the php executable at /usr/local/bin is running I was just hopeful that a conf setting somewhere could tell AGI what language/host is to be used for executing its calls. But if I can rely on the #! setting in the file, that's good enough for me. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need case study for Open Source presentation
Hi there, I have been asked by a industry group of University educators leaders to present a session to them at an upcoming conference on Open Source philosophies. I've pretty much completed my presentation outline and notes, and I'd like to include any case studies regarding the benefits (both financial and flexibility) that any medium to large organization may have experienced with using open source technologies. I have case studies for areas such as databases and web applications. What would really be ideal is if someone who has completed a relatively large scale Asterisk deployment could share some time with me to explain how it went, and now that they are using Asterisk what benefits their organization is experiencing from it. It would be even better if the organization was a university, college, or some member of the education industry, but that is not absolutely critical. If anyone has such a 'war story' that they are willing to share with me, and I can keep as much of it anonymous as possible (although I'd really like to cite the organization name that they are referring to for credibility), please feel free to email me off-list at my...@techsol.org I'll do my utmost to keep as much information private if you require it, but it would really help me 'evangelize' Asterisk into some very large and prestigious educational institutions (worldwide) who will be at this annual conference I am speaking at. Thanks and I look forward to any responses. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan strategy suggestions needed
I have a new Asterisk system going into production next week and I'm a bit stumped as to the best way to handle the Dialplans for it. The Asterisk system is replacing 4 separate PSTN lines with both SIP PSTN inputs. The setting up of the dial plan is giving me some design headaches, which probably means I'm missing something obvious and doing this the hard way. I have separate entire phone 'systems' for each incoming DID. For example, this one system will handle 4 separate incoming DIDs. The first (let's call it Company A) takes a call, plays a menu, gets input from the caller, directs them to the appropriate extension (ie. Press 1 for sales, 2 for support, etc.). The second DID has a similar setup with a menu, but its for an entirely different company. The incoming extension number I'm getting for this will be entirely different, therefore will be the definition of this phone menu structure. The others are simply numbers that go through a 'Time of Day' check and then simply forward to an extension. I have it all working fine by using: extern = 212555,1,. extern = 212555,2,. etc. for the first number, and then a second set for the second number like: extern = 2125551112,1,. extern = 2125551112,2,. The problem is when I get to the Background() command to play the sound file and get the input from the user for the menu. Since the input from the user becomes the extension, I then have a problem that my multi-faceted dialplan now gets confused with what extension applies to what menu, etc. I need to be able separate these into their own sections so that extensions won't conflict but I'm not sure how to do this. All the calls are coming in from one SIP provider, so I have only one context that I'm using because of that. I'm not sure if there is a way to create separate contexts for this and branch to them? I must be approaching this wrong. Can anyone point me in the right direction so I create something that is manageable for the future and won't have conflicts on user inputs between the systems? Thanks Myles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan strategy suggestions needed
Tim wrote:
Sure, have a top level context that inbound calls from the ITSP go into:
[from-ITSP]
exten = 2125551112,1,Goto(companya,${EXTEN},1)
exten = 212555,1,Goto(companyb,${EXTEN},1)
; then separate contexts for each company:
[companya]
extern = 212555,1,.
extern = 212555,2,.
[companyb]
extern = 2125551112,1,.
extern = 2125551112,2,.
Thanks. Yes, that is what I would have thought. But the problem is
that my ITSP has one entry in sip.conf and this goes to one context.
Should I repeat that entry for each DID that could come in from the ITSP
in sip.conf to allow this to work?
Myles
--
===
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
Scottsdale, Arizona USA
http://www.techsolusa.com
Phone +1-480-451-7440
___
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Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Jeff LaCoursiere wrote: You don't have to send the traffic back to broadvoice for outbound if you don't want or need to. Perhaps you can send the home traffic to Broadvoice and pick another carrier to send your other outbound traffic to, perhaps one that won't be so picky about your outbound CID. Thank you for this. Yes, I wasn't thinking that way at all. I suspect that I need to find a carrier that will let me have better control over CID than BroadVoice. Do you have any suggestions? I'm in Phoenix, Arizona so one that has decent network speed near us would be best. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Lyle wrote: I had this issue with Teliax. Basically with SIP, Teliax could not (or the protocol won't let you) set your outbound caller ID via Asterisk. Caller ID is set on a per account basis with Teliax when using SIP(IAX was not working well for me with Teliax). So I have two outbound pay per minute accounts with them. One for our home use and one for my business. I use 51 prefix for home outbound calls and 52 prefix for business outbound calls. Then my dialplan selects the proper account at Teliax and you get the proper caller id set. Yes, this is the same behavior I'm seeing with Broadvoice. What seems to make it even worse for this is that when callers receive my outgoing call, its showing the correct CID for the outgoing line, but the Name that is showing is always 'BroadVoice'. I asked them to have this changed to my company name, but it doesn't seem to have had any affect. I suspect that there is some master database somewhere that recipient phones lookup based on the number to get the name? If so, its not correctly identifying our company name on phones so I'm looking for alternative outbound carriers for this. The inbound, however, works well. I'm sure its not the least expensive option out there but so far its been pretty good. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
I'm pretty new to this whole Asterisk system VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home home office. My wife I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting ridiculous, so I'm moving over to a SIP carrier. I created one account for a single phone number with a SIP carrier (BroadVoice) and have it working well with my Asterisk system and one SIP phone here as a test. I have IPCop as my Firewall/IDS system and all the SIP/NAT routing stuff is working fine (now). I started the process today to get our other phone numbers moved over to BroadVoice. I checked with them regarding how this is setup and they said that what I was doing was ok, but I thought getting some 'peer review' on this wouldn't be such a bad idea so I welcome any comments, etc. on this. My approach is to have one trunk provided by the SIP provider. All numbers are allocated to that trunk (BroadVoice let me do that when I setup the number transfer). Asterisk receives an incoming call on that trunk and determines the calling number that it was requesting (not sure how to get this, but Broadvoice assured me I could). Anyway after determining what the call was destined for, I then route the call to the appropriate context in the extensions to handle it. I'm fine with setting up all the logic, flow, etc. for the calls. But here's where I'm not sure what to do. I'm getting 4 line Grandstream phones for my office and my wife's office. And an ATA adapter for the general home line. The home line will always call out using the home phone number. The office numbers, however, should change their caller ID and caller name based on which extension is pressed on the phone for the outgoing call. I can see how to do this with the Grandstream SIP phones, and have this working ok for my test phone line. Broadvoice, however, won't let me change the outgoing caller ID. Apparently they have to do this on a trunk by trunk basis. So if I want to have an outgoing call go through line 1 (let's say its ACME Inc), I want it to show 'XXX-XXX- Acme Inc' for the Caller ID. But if the call is being sent through line 2 (let's say its SMITH PROPERTY) I want it to show 'YYY-YYY- Smith Property' for the Caller ID. It looks like in order to do that, I need to purchase separate trunks for each of the outgoing lines. Does this sound right? Should I have purchased all separate trunks up front and then have the phone number transfer associated with the trunk for it? Or is this only something that will affect outgoing calls, so its not a big deal? And what about when the line is busy? How is that handled? I was on the phone yesterday when another call came in, and it came in, jumped to a different extension and then eventually went to voice mail as I didn't answer it. Will my plan to use one trunk for all incoming lines make sense here, or am I likely to get all of this mixed up with calls coming in for one business and being routed to the wrong place? Any suggestions, thoughts or critique would be greatly appreciated. Thank you wise Asterisk gurus! Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
