Re: [asterisk-users] 2000+ user Asterisk PBX
Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs Femi I would be interested in some of the replies if you wanted to continue the topic on-list... Your problem might help someone else down the line. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] bandwidth required for Asterisk running on T1
Hi, I want to estimate the amount of bandwidth required for Asterisk running on a T1 in a typical scenario. Can someone share with me any implementation experience? Thanks in advance for your input. Regards, Mark Check out http://www.asteriskguru.com/tools/bandwidth_calculator.php it should help you figure out how much bandwidth you will need. Ryan ___ -- 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] Asterisk scalability
Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6 idle cores. Is this correct? Why not? If this is correct, increasing CPU number of Asterisk server box would not increase the performance. I don't see any other process that could use other Cores (like transcoding processes, executing dialplan, etc.) Thank you for your information, -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona Carles, Asterisk is one process, but as you mentioned multi-threaded as well. Because it is multi-threaded it can run on multiple cores/CPU's at a time. I don't know the internals of Asterisk that well so I can't site specific examples, but I know that there are some scalability bottlenecks people are looking at, specifically with the IAX protocol and how the threads send/receive packets. I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Ryan ___ -- 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] asterisk-1.2.26.tar.gz Thoughts?
At 11:53 AM 1/18/2008, you wrote: Apart from the fact asterisk 1.2 is in security maintenance mode only and wont get any other bugfixes it will be ok. Please consider using 1.4 as it's the official latest stable version. Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Ira What type of Asterisk setup do you have? While my setup is not a large commercial setup I have seen asterisk 1.4 with a few calls going through it at once last for weeks if not months before it was restarted. Just curious. ___ -- 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] Using XML for configuration management, single-source-of-truth, etc.
Tilghman Lesher wrote: On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. Well, after hand-coding HTML and SGML for 15+ years, XML isn't all that much of a stretch. More to the point though, there are some excellent schema-driven configuration managers for XML, so you wouldn't have to edit the files by hand. -Philip Can these configuration managers run from a command line? Or do they require a graphical environment? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH} ${FUTURETIME})
exten = s,n,Wait(0.01)
exten = s,n,EndWhile()
exten = s,n,Play(beep)
exten = s,n,Hangup()
and hating myself for it (my Asterisk runs on a 500MHz Geode LX).
So I decided it would be useful (in general, and educational for me in
particular) to write a WaitUntil() application instead.
Well, I've done that.
I was going to file a bug and then post a fix to get their feature in,
but the Bug guidelines seem to be pretty clear that this is not the way
to go.
So, I'm posting here instead.
The example to paste into the documentation or extensions.conf is:
[popcorn]
exten = s,1,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages is there's no pause before the synchronizing beep.
exten = s,n,Set(FUTURETIME=$[${EPOCH} + 11])
exten = s,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten = s,n,SayPhonetic(z)
exten = s,n,SayUnixTime(${FUTURETIME},,HNS)
exten = s,n,Playback(local)
exten = s,n,WaitUntil(${FUTURETIME})
exten = s,n,Playback(beep)
exten = s,n,Return()
I invoke it as:
exten = 712,1,Gosub(popcorn,s,1)
exten = 712,n,Hangup()
And lastly, attached is the source for app_waituntil.c.
It's fairly straightforward, and not very big.
But hopefully useful.
Oh, before I forget: it does require the recording of one additional
phrase,
either local or localtime. I've used local in my example above.
And
I read out the time first as GMT/UT (because I travel a lot), and then in
the
timezone of my PBX...
-Philip
Philip,
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
Ryan
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Re: [asterisk-users] New feature: calling all bug marshals
In article [EMAIL PROTECTED], Ryan Burke [EMAIL PROTECTED] wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? It's just doing a standard round to nearest integer division, by adding half the divisor to the dividend before dividing. Without that, you just get round down instead. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org I see, sorry it was a brain fart... Thanks! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Site Extensions That Would Show As In-Use?
Dovid B wrote: Are you trying to get FOP to monitor the SIP account that you are using to dial the cell phone on ? The SIP extension, yes. So, as long as a call that has been forwarded to that cell phone is still in progress, that extension should still show busy. Thanks again, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-Site Extensions That Would Show As In-Use?
Hello, list! I'd like to create an extension that points to an offsite location (a number on the PSTN), the purpose of which would be to see if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom extension with the destination SIP/TrunkName/NXXNXX, the calls transfer fine but don't show as busy using the Flash Operator Panel (as an example). Any thoughts? Thanks in advance, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Waiting before executing System command
Hello, Moses!
At 09:20 PM 27/10/2006, you wrote:
what about
exten = h,n,System(mycommand /some/file /some/other/dir/)
Where mycommand is your custom shell script to sleep before moving the file.
That would work, but I'm trying to avoid kludges like that. Hence my
question about doing it entirely within the dialplan.
Any ideas?
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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[asterisk-users] Waiting before executing System command
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So instead I tried:
exten = h,n,System(sleep 5; mv /tmp/${CALLFILENAME}
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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Re: [asterisk-users] Modifying SIP Stack
I don't have access to the sip code right now, but from past network code I've writen you could try this: inet_aton(192.168.1.10, p-sin.sin_addr); If they are just wrapping the struct sockaddr_in as sin in p. Worth a try... Ryan Hello list, I am trying to include a new message after I receive a Register in chan_sip, at the beginning I would like to forward the same message to a fixed IP address, I have seen that fileds like p-sa.sin_addr and p-sin.sin_addr have to be with the IP address, but I am not sure about how to force these fields to be the destination that I want (192.168.1.10)... somebody knows how to do it? do you know where can I fond information about the p structure in asterisk and how to modify it manually? Thanks and Rgds, German ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why is this happening?
Do me a favor and try running netstat -aplntu | grep asterisk and see what ports are actually being used. Are you connected to another ITSP? If so then that may be the local port of that connection... just an idea, i don't have Asterisk access right now to double check. Ryan On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote: Why is it running on port 1207? because Asterisk is listening on port 4569 and when a connection comes in, it as handed to another port so it can continue listening on port 4569. Otherwise you would only be handling 1 connection at a time. Pretty basic networking stuff I think :c) Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test to list
Sorry, just checking if my mail is working. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
I would like to know how you got Asterisk to function with 2500 SIP registrations. Did you have qualify enabled? Yes, qualify was enabled, using the standard length of qualification period between checks. Very few accounts had custom qualify settings. What about the 500 simultaneous calls? How many SQL hits were you doing (all said and done). Any performance logs from the SQL server? I can't believe you got all this running on one box! You have to remember, 500 simultaneous calls is not the same as something like 20 calls per second. some of those calls may have been quite long, and once the call's been placed, there's no database work being done until the call ends. I wish I had statistics from that setup, but I don't, we spent so much time implementing new features and chasing down problems caused by using a pre-RTA version of Asterisk with a patched in RTA setup. -- S McGowan VoIP Consultant [EMAIL PROTECTED] S McGowan, I don't know if you missed my question (from the slew of questions you've received and answered), but I was wondering about transcoding and PSTN channels. What kind of codecs were used and was there any transcoding happening? Was this box only responsible for VoIP-to-VoIP calls or was there also PSTN trunks as well? Again, I'm amazed by this example since it seems to be way over what anyone else normally reports as usable. Thanks again, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
Benjamin Jacob wrote: Rushowr wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Date: Tue, 19 Sep 2006 05:52:52 -0500 Marco Mouta wrote: Hi all, I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions? Can i trust in a solution only with Asterisk to make all this install? Please help me with your experience on this kind of asterisk solutions. I've googled and read about asterisk at large scale solutions, but still in doubt. http://www.voip-info.org/wiki-Asterisk+at+large -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In my experience, yes you can use *just* asterisk for the implementation of a large scale setup, you just better be sure you've planned it out well. I've set up a few large scale Asterisk implementations, covering more than 1K users on a single box. And that was in 2005 using trunk. There were problems, but all in all it was (and is, for the former client) not a bad implementation. If you're just looking at a large PBX install, you're definitely fine with a well planned system. Just my $0.02, not to be taken as a guarantee ;-) You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this is the one!!)??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers and realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a peak (that I recall) of around 500 concurrent calls. -- S McGowan VoIP Consultant [EMAIL PROTECTED] Can you explain your design in a little more detail? What kind of hardware did you use to get over 1k users on a single box and 500 concurrent calls? Sounds like a very interesting medium-large scale implementation that others could learn from. thanks, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?
- Original Message - From: Rushowr [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 19, 2006 10:38 AM Subject: Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ? Thanks for the info. So it was really just one server that handled 2.5k user registrations and up to 500 concurrent calls? Do you remember anything about the codecs? Was there any transcoding done, music on hold, queues, etc? Usually for a dual Xeon 3Ghz people say they get about 250 concurrent calls and maybe 1k users registered before things start acting flaky. I really appreciate the info. I'm looking forward to hearing about your current project when you get a chance to write it up. Thanks again, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is anybody using XEN in conjunction with Asteriskand/or Openser?
I had that running in my small home Asterisk setup for a while, however I had to move back to an older kernel to support a SATA driver (mv_sata). It looks like as of late 2.6.16 and 2.6.17 they've updated the driver so in a week or so I am going to try builind it again and create a DomU with Asterisk with a X100P clone FXO port. I can't remember all I did before, but I'll be sure to post my experiences in the coming weeks. Ryan - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, June 24, 2006 2:54 AM Subject: [Asterisk-Users] Is anybody using XEN in conjunction with Asteriskand/or Openser? Is anybody using XEN in conjunction with Asterisk and/or Openser? I would like to get some info about such an environment and experience reports. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Custom extension halting execution upon caller hanging up
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten = 2,8,Monitor(wav,${TIMESTAMP})
exten = 2,9,Dial(SIP/Provider/8005551212)
exten = 2,10,Macro(record-cleanup)
If the caller hangs up before the callee does, execution of the
custom extension halts and does not continue to priority 10
(record-cleanup), where sox is used to reverse the audio files and
then mix them then reverse them again so they'll be in sync (since
inbound audio only starts from call-answered but outbound audio
starts from the beginning of ringback).
Asterisk provides this debug output to the console (internal
extension 101 is the caller):
-- Called Provider/8005551212
-- SIP/Provider-993d is making progress passing it to SIP/101-1666
-- SIP/Provider-993d answered SIP/101-1666
The call proceeds normally, but then Asterisk spits this out the
moment the caller hangs up first:
== Spawn extension (custom-extension, 2, 9) exited non-zero on
'SIP/101-1666'
How can I prevent the extension from bailing before I have a chance
to clean up the recording in priority 10?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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[Asterisk-Users] Custom Extension halting execution upon caller hanging up
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten = 2,8,Monitor(wav,${TIMESTAMP})
exten = 2,9,Dial(SIP/Provider/8005551212)
exten = 2,10,Macro(record-cleanup)
If the caller hangs up before the callee does, execution of the
custom extension halts and does not continue to priority 10
(record-cleanup), where sox is used to reverse the audio files and
then mix them then reverse them again so they'll be in sync (since
inbound audio only starts from call-answered but outbound audio
starts from the beginning of ringback).
Asterisk provides this debug output to the console (internal
extension 101 is the caller):
-- Called Provider/8005551212
-- SIP/Provider-993d is making progress passing it to SIP/101-1666
-- SIP/Provider-993d answered SIP/101-1666
The call proceeds normally, but then Asterisk spits this out the
moment the caller hangs up first:
== Spawn extension (custom-extension, 2, 9) exited non-zero on
'SIP/101-1666'
How can I prevent the extension from bailing before I have a chance
to clean up the recording?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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[Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable
Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 username: anonymous password: your email address -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement: New Texas User Group formed
Sounds great. I'm just a home user of Asterisk, but I love the product and have recommended it to alot of other people. Let us know when the site is up. Ryan - Original Message - From: Bruce Reeves To: asterisk-users@lists.digium.com Sent: Thursday, April 13, 2006 12:30 AM Subject: [Asterisk-Users] Announcement: New Texas User Group formed In an effort to bring Asterisk Users from across the state of Texas together, the Texas Asterisk Users Group has been formed. The goal is to help Asterisk users meet other is their area and to help spread the word about the Asterisk community. I anticipate regional meetings of members and look forward to all of our members being able to attend the Astricon Event in Dallas in October. I invite all of the Asterisk users in Texas to visit our web new website, still under development, and join our group and our mailing list. I am currently taking suggestions on the format and backend of the website and look forward to helping build a stronger asterisk community in Texas. -- BruceNortex Networks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Texas User Group
I'm interested but I'm in the Dallas area. Are there any in the Dallas area anyone knows of? Ryan - Original Message - From: Bruce Reeves To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 10, 2006 12:51 PM Subject: [Asterisk-Users] Texas User Group I am wondering if any of the Texas user groups have members in the North West part of the state. I am in the Amarillo area and would like to find some othere in this area, maybe even start a user group in this area.-- Bruce ReevesNortex Networks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
Erik: At 01:17 PM 03/22/2006, you wrote: On 3/22/06, Andrew D Kirch [EMAIL PROTECTED] wrote: Andrew D Kirch Indianapolis, United States snip Well if that isn't one of the most bizarre emails I've seen come across this list. It's a spoof of a typical Nigerian 419 scam email. Rather well done, too. :) Thanks for the laugh, Andrew! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7970 SIP Firmware; SIP 8.2 for 7940/7960
Hello! I'm hearing about this 7970 SIP firmware. I'm a Cisco Registered Partner with full access to the Cisco Software Center, and yet I can't find it. Can someone enlighten me as to where to get it? Is it also available/applicable to the 7971G-GE? Did you know that on March 10, SIP 8.2 was released for the 7940/7960? Has anyone tried it yet? If so, what are people's opinions? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] courtesy message calling mobile phones
At 12:07 PM 02/27/2006, you wrote: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? My knowledge of SS7 is limited, but this has to do with opening the audio path before a call-answered event (which never comes), or even before a call-alerting event. This is also the case where a SIT is generated, and a message like the number you have reached is not in service is played for those not hardcore enough to know the specific error from the sound of the SIT alone. :) -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie config help? Wellgate 3701a
Hello, Martin! At 02:50 AM 02/26/2006, you wrote: I got my new Welltech 3701a, 1FXS,1FXO gateway. If you do give up with it (isn't Engrish documentation fun?), you may wish to take a look at the Sipura SPA-3000. I have one but haven't put it to use yet. I've heard *many* good things about it, though! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Web-Based Voicemail?
Hello, list! After Googling and checking out the voip-info wiki, I haven't had much luck in locating a decent web-based voicemail system for Asterisk to check your VM while you're away from the office without using a phone. Can anyone make any recommendations for such packages/applications? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Over WiFi
Hello, Dumpexec! At 12:35 PM 02/26/2006, you wrote: Is there a sort of high grade cat5 cable that can propagate signals for up to 1Km? No. The standard is 100m per leg, maximum, even with STP (shielded twisted-pair) cable. You could go to multimode fiber to get 2km, but you'd have to find another way to power your device. Sorry! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepaid / postpaid solution
At 05:03 PM 02/26/2006, you wrote: I want to match the user from the users callerid. All users have DIDs. You probably shouldn't do that for security reasons -- rather, match them according to the SIP username/password pair they provide when they register. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, Steve! At 03:55 AM 02/21/2006, you wrote: ztdummy was only used for timing. Linux 2.6 provides this function in the kernel and I assume Solaris already has timing functions there. Page 36 of Asterisk: The Future Of Telephony (O'Reilly Press) states that you either require a Digium PCI card to provide clocking, or ztdummy if you lack the PCI hardware required to provide timing. It goes on to mention that a UHCI USB controller was required pre-2.6 but now that there's a 1kHz clocking source in the kernel, ztdummy will attach to that instead, thus eliminating the requirement for the UHCI USB controller. While it doesn't explicity say so, it seems to very strongly imply that either a PCI card or ztdummy are *required* for some Asterisk functionality (namely music-on-hold and conferencing, apparently). Is this actually not the case? Just for reference, here's the section in question, verbatim (copy-and-paste from the PDF): The ztdummy Driver In Asterisk, certain applications and features require a timing device in order to operate (Asterisk wont even compile them if no timing device is found). All Digium PCI hardware provides a 1-kHz timing interface. If you lack the PCI hardware required to provide timing, the ztdummy driver can be used as a timing device. On Linux 2.4 kernel based distributions, ztdummy must use the clocking provided by the UHCI USB controller. The driver looks to see that the usb-uhci module is loaded and that the kernel version is at least 2.4.5. Older kernel versions are incompatible with ztdummy. On a 2.6 kernelbased distribution, ztdummy does not require the use of the USB controller. (As of v2.6.0, the kernel now provides 1-kHz timing with which the driver can interface; thus, the USB controller hardware requirement is no longer necessary.) The default Makefile configuration does not create ztdummy. To compile ztdummy, you must remove a comment marker from the Makefile. Open it in your favorite text editor and look for the following line: MODULES=zaptel tor2 torisa wcusb wcfxo wctdm \ ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy Remove the hash* (#) symbol from in front of ztdummy, save the file, and compile Zaptel as usual. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
Hello, Mitchel! At 07:41 AM 02/21/2006, you wrote: I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! As I understand it, PCI-X is fully backwards-compatible with PCI (as in the presence of a PCI card on a PCI-X bus will cause that bus to drop back to regular PCI mode). If you want something super-reliable which can run Linux, Solaris, or Windows, and you require three PCI slots, this may interest you: http://www.sun.com/servers/entry/x4200/ (Click on the Gallery link for pretty pictures.) I'm seriously considering two X2100s (because I don't need four disks or any PCI cards): http://www.sun.com/servers/entry/x2100/ These boxes will run Solaris, Linux, or (ack) Windows, and their remote monitoring/management support is second to none. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Download Asterisk: The Future Of Telephony [More Info]
Hello, Bob! At 01:32 PM 02/21/2006, you wrote: Speaking of this book, where can I get it? Obviously I can read the pdf, but I lack the facility to print it in any usable fashion. The labor and materials that I have spent on trying to print it thus far probably outweighs the cost of the silly thing. Is it only available online, or do you think Barnes and Noble, Borders, etc might have it? Oh, I wouldn't print the whole thing; the price of the paper copy doesn't make it cost-effective to run one off... unless you happen to work at a place with a nice laser printer and a spiral-binding machine, I guess! Any reputable book seller should be able to order it by its ISBN (0596009623). I bought my paper copy from Amazon, and had it in a week. It *is* a real book -- the PDF that was released is (most of) exactly what went to the book printing company -- the markings in the corners are alignment marks, and the vertical and horizontal lines in the margins are the cut marks for binding. The table of contents and index are missing, probably because they're fairly useless in a file you can do full-text searches on, and also probably to make counterfeiters actually have to do some work. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, Mark! At 06:33 AM 02/20/2006, you wrote: Please forgive the question, but what is the rationale behind using Solaris over Linux as an asterisk hosting platform? Because of a few reasons, actually: (1) The remote hardware management options available for the X2100 work better (or only, I'm not sure which) under Solaris, and they seem to *really* kick ass. Plus, being Sun-engineered, the X2100 should keep working until it's completely obsolete, and then some. (2) I know someone who knows Solaris inside-out and backwards, blindfolded, while hung upside-down, and codes Bourne shell and C in his sleep; this is vaguely reminiscent of www.chucknorrisfacts.com. I'm quite sure this will come in handy when (not if) something breaks, giving him the opportunity to make some money and giving me the opportunity to reduce my downtime. :) (3) I'd like to learn Solaris, and being SysV-based like Linux, it shouldn't be too much of a stretch. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello all, I really appreciate the replies I've gotten about this so far (especially the support for wanting to run it on Solaris!). The core issue seems to have been missed, though -- is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? This probably comes down to a few issues: - Is ztdummy (a component of Zaptel) *really* required for MoH and conferencing support? - I've heard rumblings about zaprtc being a potential replacement. Is it a *real* replacement? Will it work on Solaris 10? If not, what will? - I *know* people have got to be running Asterisk on Solaris 10 (but I don't know who they are, unfortunately!). If you happen to be a member of that esteemed clique, could you please let me know how you got ztdummy working, or what you used as a replacement? I really don't see people going without MoH and conferencing in a real setup. Thanks again! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada Date: Sun, 19 Feb 2006 23:45:01 -0500 To: asterisk-users@lists.digium.com From: Alexander Burke [EMAIL PROTECTED] Subject: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100) Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge 2850
Hello, Klaus! At 06:23 PM 02/20/2006, you wrote: Both riser cards only have 64 Bit PCI slots. I think 64 bit is always 3.3 Volt - isn't it? Nope! http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Grab a copy of Asterisk: The Future of Telephony from the link above, and open it to page 16 (PDF page 34 or thereabouts). There's a great illustration of the possibilities. Plus, it's a great book to have on hand; I bought a paper copy before I knew it was available online, but I still would have bought it, because it's nice to have. Kudos to O'Reilly for setting it free, too! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Hello, Marc! At 06:24 PM 02/20/2006, you wrote: Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P's in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration According to Asterisk: The Future Of Telephony: IRQ latency Interrupt request (IRQ) latency is basically the delay between the moment a peripheral card (such as a telephone interface card) requests that the CPU stop what it's doing and the moment when the CPU actually responds and is ready to handle the task. Asterisk's peripherals (especially the Zaptel cards) are extremely intolerant of IRQ latency. Linux has historically had problems with its ability to service IRQs quickly; this problem has caused enough trouble for audio developers that several patches have been created to address this shortcoming. So far, there has been some mild controversy over how to incorporate these patches into the Linux kernel. Because the Digium cards require so much, it is generally recommended that only one Digium card be run in a system. If you require more connectivity than a single card can provide, either replace your existing card with one of higher density, or add another server to your environment.* * Many people report that Sangoma cards are more robust when it comes to dealing with unpredictable motherboard chipsets, and thus can handle sharing motherboard IRQ resources. Regardless, it is still worth considering using multiple servers, as the redundancy that can be gained from this strategy can quickly offset the cost. -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Download Asterisk: The Future Of Telephony
Hello, list! I'm hosting a mirror of the book Asterisk: The Future Of Telephony by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the filesize with Acrobat's Reduce File Size tool (while still maintaining compatibility with Acrobat 4.0, apparently). I bought a paper copy before I knew the book was available online, but it's good enough that even had I known it was available online, I still would have bought it on paper. You're welcome to download it and keep it on hand -- it makes for EXCELLENT reading: http://www.alexburke.ca/asterisk-tfot.pdf -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Marc: At 06:24 PM 02/20/2006, you wrote: I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration This bad boy might be what you need: http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM2400Ptab=details If not, consider an external channel bank: http://www.voipsupply.com/product_info.php?products_id=868 http://www.voipsupply.com/product_info.php?products_id=781 It would be great if you could let the list know which route you take, and the success (or lack thereof) that you have with it! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Download Asterisk: The Future Of Telephony [More Info]
One thing I forgot to mention: I also cropped the registration and cut marks off the sides of the pages. If you want the uncropped version, get: http://www.alexburke.ca/asterisk-tfot-uncropped.pdf Sorry about the excessive noise, but I figured I should mention this. Date: Mon, 20 Feb 2006 18:55:50 -0500 To: asterisk-users@lists.digium.com From: Alexander Burke [EMAIL PROTECTED] Subject: Download Asterisk: The Future Of Telephony Hello, list! I'm hosting a mirror of the book Asterisk: The Future Of Telephony by O'Reilly Press, published under the Creative Commons license; I believe this license allows me to do this, but if I'm mistaken, please let me know. I've taken the liberty of fixing the page numbers so Acrobat is now aware of the correct number of each page, and shrinking the filesize with Acrobat's Reduce File Size tool (while still maintaining compatibility with Acrobat 4.0, apparently). I bought a paper copy before I knew the book was available online, but it's good enough that even had I known it was available online, I still would have bought it on paper. You're welcome to download it and keep it on hand -- it makes for EXCELLENT reading: http://www.alexburke.ca/asterisk-tfot.pdf -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the timing-source component of Zaptel, is required for the music-on-hold and conferencing functions of Asterisk to function. So, with this in mind, is there any way to run a complete Asterisk solution on Solaris 10 (including music-on-hold and conferencing)? If so, how? Thanks in advance! -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speech playback getting cut off
List, I purchased an SPA-1001 from voipsupply.com (which was great.. no hassle), but now anytime there is silence and then Playback() sound files the first half second gets cut off. For example, when I login to the VoiceMailMain() function I don't hear "Password",I hear "-assword". Then after I login I hear "- have no messages..." instead of "YOU have no messages...". Has anyone experienced this? I tried turning off the echo cancellation stuff in the SPA-1001 but that didn't help. I can always add a "Wait(.5)" before the "VoiceMailMain()" but doesn't help once I'm in the application.I'm sure there is somthing I'm missing, probably in the SPA-1001 config, but I'm hoping for some help from the list. Thanks in advance. Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech playback getting cut off
Any ideas? I appreciate hte help. Ryan - Original Message - From: Ryan Burke To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 7:58 PM Subject: [Asterisk-Users] Speech playback getting cut off List, I purchased an SPA-1001 from voipsupply.com (which was great.. no hassle), but now anytime there is silence and then Playback() sound files the first half second gets cut off. For example, when I login to the VoiceMailMain() function I don't hear "Password",I hear "-assword". Then after I login I hear "- have no messages..." instead of "YOU have no messages...". Has anyone experienced this? I tried turning off the echo cancellation stuff in the SPA-1001 but that didn't help. I can always add a "Wait(.5)" before the "VoiceMailMain()" but doesn't help once I'm in the application.I'm sure there is somthing I'm missing, probably in the SPA-1001 config, but I'm hoping for some help from the list. Thanks in advance. Ryan ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Testing List (JUST A TEST)
Sorry, I haven't received a message in a few hours, just testing to see if it is alive. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk down because of cdr
Buffers don't have to be in memory. My suggestion on the solution would be to buffer the CDR info into a backup file based database (configurable filename/path) on the local filesystem (or NFS mounted system for redundancy) and then when the SQL database connection is restored then it spends a second dumping the buffered CDR info from the file into the database and erases the file (or empties it). This is would be a fairly solid buffering solution. Of course you better get that SQL database back online ASAP or else there is the issue of running on of diskspace on heavily used systems. Just an idea, doesn't seem that hard but I know I don't have time to implement it now. Ryan Buffer! For how long? How big of a buffer? If I can buffer 10-20 calls that might work if I have a light use PBX but 100-2000 buffered calls may not hold a busy PBX. OK so make it configurable, With any luck you won't know how much to put so you will allocate more than you need, using more memory for a single senario. My solution, make sure your DB is stable. I would rather put my effort in building a better solution than counting on insurance to bail me out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, January 17, 2006 12:38 PM To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk down because of cdr Dov Bigio a écrit : Ok.. but I don't use Real Time at all. I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages or at least just logged, but without stopping. What would be even nicer would be for * to buffer it for a while before it starts dropping cdrs... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco/asterisk interop issues?
hi, i have an issue that when making a call from a SIP phone going as follows: phone -- asterisk -- cisco(192.168.0.1) -- terminating voip platform(10.0.0.1) i get the cisco sending up an invite to the voip platform followed directly with a CANCEL message, as follows: Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4 Remote-Party-ID: device sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off From: device sip:[EMAIL PROTECTED];tag=B2A336CC-413 To: sip:[EMAIL PROTECTED] Date: Thu, 05 Jan 2006 15:09:08 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer,resource-priority Min-SE: 1800 Cisco-Guid: 227404060-2100564442-3154699218-4120052929 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REG ISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1136473748 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 285 Jan 5 15:09:10.642: //-1//SIP/Msg/ccsipDisplayMsg: Sent: CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4 From: device sip:[EMAIL PROTECTED];tag=B2A336CC-413 to: sip:[EMAIL PROTECTED] Date: Thu, 05 Jan 2006 15:09:08 GMT Call-ID: [EMAIL PROTECTED] CSeq: 101 CANCEL Max-Forwards: 70 Timestamp: 1136473750 Reason: Q.850;cause=0 Content-Length: 0 the asterisk reports the following: -- Executing Dial(SIP/200-c5c4, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/192.168.0.1-a928 is making progress passing it to SIP/200-c5c4 -- Got SIP response 500 Internal Server Error back from 192.168.0.1 -- SIP/192.168.0.1-a928 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) if i send it as follows: phone -- asterisk -- cisco(192.168.0.1) -- pstn all is good and call is processed normally. any help would be appreciated.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound sip calls on asterisk
hi, i would like all my calls originating from asterisk users bound for external to route to one destination, a session border controller. protocol used is sip. i have edited extensions_custom.conf with: exten = _.,1,dial(sip/[EMAIL PROTECTED]) would this be correct to send any calls from internal to the x.x.x.x ip? i get this from the cli: == Spawn extension (from-external. then it just times out and dumps the calls? the phone used is a cisco7960 using pos3-07-5-00 any help appreciated... :) james ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-1001 question
Asterisk-Users, Is anyone out there using the SPA-1001 for integrating existing analog phones into a VoIP setup? My question has to do with the MWI. From the datasheet it says that it provides MWI Tones, and then that it provides Visual MWL via FSK. What does via FSK mean? My exsting phone has an answering machine built in and I am debating using Asterisk as the Voicemail, or just the exsting answering machine. Any comments or insight into the SPA-1001 would be appreciated. Thanks, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FC3 or FC1 (or something else?)
I'm currently using CentOS 4.2 in my home install on a P3-600/512MB/40GB HDD with a X100P clone and it works great. Using Asterisk 1.2.1. Ryan Any thoughts on CentOS-4.2? It is based on RHEL4 update2. It has the 2.6 Kernel. I am currently using CentOS-3.5, which is based on RHEL3 update5, with no issues. The Kernel is 2.4.21-32.0.1.ELsmp. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Brett, Gary [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi I wish to install asterisk 1.2 (the latest tar.gz from the site not the CVS version) on an HP box with a TE110P (single port E1/T1) My question is which OS would be preferred in this configuration Fedora Core 1 or Fedora Core 3, and are there any install guides out there that are recent enough for asterisk 1.2 I am also open to suggestions for other Operating Systems if any of you feel that FC1/3 are not the best for the job, my only definates are that I use the latest tar.gz from the asterisk.org website not the CVS and also that I will be using the TE110p Any help would be greatly appreciated Gary ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialling out with clone X100P board
I had the same problem at first. Try adding a w or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged message, and the call has some strange charateristics.
If I call 123, I can connect to and hear the time clock provided by BT
(I'm in the UK) Is this 'audio before answer'?)
If I call any other external number, eg my cellphone, it never rings,
and after 30 secs or so the Zap channel hangs up.
I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
What should I be looking for in my setup?
Many thanks, and happy Christmas to all.
Roger
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[Asterisk-Users] Teliax billing question
Teliax users, I have a couple questions about Teliax, just hopeing some current customers might shed some light on them. How reliable is a toll-free number from Teliax? Has anyone had any problems with it? The Pay as you go plan has a Billing of 60/1, what does that mean? My guess is 60 seconds minimum (does this apply for incoming AND outgoing, or just outgoing) and a period of 1 bill per month? For a total bill per month, it would be $.02 per outgoing minute (with a minimum of 60 seconds per call)and $.029/minute for incoming toll free (don't know if minimum time applies here) plus $4.99 for a toll-free number, right? Is there any other charges because of the toll free number? Thanks for your help, Ryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Teliax billing question
wolfgang, Thanks for the heads up. I'm hoping to get some feedback from Teliax toll-free customers and see if they would recommend the service. Plus I have those few questions on billing. Thanks again, Ryan - Original Message - From: Wolfgang S. Rupprecht [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 17, 2005 7:05 PM Subject: [Asterisk-Users] Re: Teliax billing question Ryan Burke [EMAIL PROTECTED] writes: Is there any other charges because of the toll free number? I was toying with the idea of getting an 800 number too, but the issue of a substantial per call fee for pay-phones calls has me worried. Hopefully someone here can clarify what the deal is there. I've seen numbers quoted as high as a 60-cents for the payphone settlement. from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html Watch Out for New 1-800 Number Scam - An old scam may be cropping up again for consumers with personal 1-800 numbers. Most long distance companies charge subscribers a per-call fee for calls placed from a payphone to a residential 1-800 number. This fee is then sent back to the owner of the payphone. While this arrangement is perfectly legitimate, in 2002, scammers in Berkeley, California found a way to take advantage of the system. They set up a phony payphone company and connect a bank of payphones to an automatic dialer. The dialer then randomly dialed 1-800 numbers until it hit a residential toll-free number. When the call is picked up, the scammer pocketed the 24¢ fee. Thanks to the auto-dialer, they could quickly rack up profits from the scam. By the time the operation was shut down by police, they had netted almost a half million dollars. Reports of a similar scam are coming in and consumers with residential 800 numbers are urged to check their April and May long distance bills for mysterious one-minute phone calls from Denver, Colorado. If you find such a call, be sure to contact your phone company. For more information on this scam, click herei. (Thanks to ConsumerWorld.org for this tip.) WIRELESS WATCH -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Teliax billing question
Rich, Thanks for your feedback. Sounds like what I was looking for. I think I'll sign up tonight! Thanks, Ryan - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 17, 2005 8:30 PM Subject: Re: [Asterisk-Users] Re: Teliax billing question Is there any other charges because of the toll free number? I was toying with the idea of getting an 800 number too, but the issue of a substantial per call fee for pay-phones calls has me worried. Hopefully someone here can clarify what the deal is there. I've seen numbers quoted as high as a 60-cents for the payphone settlement. from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html The scam isn't new, and its certainly not limited to home 800 numbers. The same basic principles were used by some of the 900 number folks a few years ago as well. Hell, even the local telephone companies are doing weird billings, and the average home owner never bothers to read the details to even recognize it. It's really no different then any other invoice; pay attention to what you are being invoiced. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )
I was reading the pdf and found a command that might be of some use: Prefix() ex. exten = 8661234567,1,Prefix(1) exten = 18661234567,1,NoOp() exten = 18661234567,2,Goto(800-in) After the Prefix() the the next exten is n+1 (which is 2 in this example) with the new extension (which is 18661234567 instead of 8661234567 which was originally dialed). Personally I think this is a bit more elegant than having a bunch of Goto's for each extension, but they do the same thing. Personal preference I guess. Ryan - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 10, 2005 7:08 PM Subject: Re: [Asterisk-Users] extensions and regular expressions ( probablyan easy question ) Or, just do... exten = 18661234567,1,Goto(800-in) exten = 8661234567,1,Goto(800-in) It's kind of tough to truly understand what you are trying to accomplish (or ask for). Apparently you've got something more in mind that words are making it through the list. Reading between the lines, it would appear from the 800-in that calls are coming in from some external source, and you trying to do something with them. Can you be a little more explicit. Hi Dan, Thanks for the info, but what I'm after is the ability to match a digit/character 0 or 1 times at the beginning of the string. If I'm reading your example right, it'll match anything starting with 866, which doesn't work for me. I am trying to match: 18661234567 and 8661234567 Sean ps: The pdf doesn't have a good explaination of this either, although it occurs to me that this might not be possible with * if I'm having such a hard time finding it. Daniel Wright wrote: Sean Kennedy wrote: Hi all, I'm having a hard time finding information related to the regular expressions that can be used in a dialplan, specifically as an extension. For example, I have an 800 number which I'd like to jump directly to if my users dial it, instead of going over my pstn termination. Currently, it looks like this: exten = 8661234567,1,Goto(800-in) However, I'd like 1866123456 to match as well. I can't find in the wiki or sample configs how to say match this 0 or 1 times. Can anybody provide a link that would go over this? Again, I've been digging through the wiki, but I seem to be missing it. Thanks Sean You could do it like this: exten = _866.,1,GoTo(800-in) The period means match one or more characters. You can find reference to expressions and how they work in this pdf book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Dial Failover
Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In the event it gets no TRYING response to a dial command within a specified timeout it should return control and flag an error. If on the other hand it does get a TRYING response (and maybe a RINGING too) it should continue to wait until the 20s has expired. I can't use dynamic DNS (ie putting two A records for a hostname in DNS) because Asterisk reads the extensions.conf on startup and also seems to cache what the host maps to on startup. Subsequent calls to the host always return the same IP address. But... in general... how have people implemented this? Help appreciated! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Dial Failover
WEll I personally have not implemented a Linux-HA cluster mainly because I don't have the resources to do so. I study Asterisk purley as a hobby (nerd.. yeahI know) because it is an awesome OSS product. Anyways, after some searching around I think it would not be TOO difficult to implement a resource failover using a combination of Linux HA (linux-ha.org), Mon (www.kernel.org/software/mon) and SIPp (sipp.sourceforge.net). Linux HA supports failover on resources or machine based failures(inherently if a machine goes down all the resources will be gone too). However Linux HA just provides the interface to say which machine is the Active machine in the cluster for a specific resource (aka Asterisk). That is where Mon would come in. Mon montiors services on a machine and can be configured to react if a service fails the periodic test. But Mon does not have a per-defined monitor script for Asterisk. That is where SIPp would come in. You could create a Mon script that calls SIPp and looks for the return code after a number of calls through the server. SIPp will return a 0 if all the calls succeeded. That way in your Mon script if SIPp returned anything other than a 0 then register it as a failure. Once a failure occurs you can configure Mon to switch the active Asterisk server using the Linux HA functionality. Like I said, there is turn key way of doing this, but it looks like a good little project for the wiki? Maybe I'll start working on this in my spare time, I just need to get some time to play with the different components. There are a few more logicistical things that would have to be taken care, mainly anything file realted, but that could be alleviated with some kind of remote mounted filesystem. Hope this helps, Ryan Yes, that's a great question. I'm wondering the same thing. Can these heartbeat apps monitor applications as well as network connectivity? The heartbeat utility at www.linux-ha.org talks about monitoring some standard apps like web servers and such but isn't clear about other apps... like Asterisk or SER. -Original Message- From: John Cianfarani [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 8:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Ryan/Jonathan, Couple quick questions regarding your setup? Do you operate this in a strictly master/slave setup? Do you have anything(mon/ha's internal status/monitor options) that actually monitors the asterisk process (to determine if it is hung). Or is it only with total box failure to you fail over? Do you use something to sync config/vm/cdr? Rsync/unison? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In
Re: [Asterisk-Users] Wait for X rings before answering?
seconds I realize that it's a timeout but what's implicit in that is that Asterisk can't detect # of rings just the amount of time spent ringing? I have been looking at the reference manual on asteriskguru.com. They say it's a timeout but they don't indicate the units. Is it milliseconds, microseconds or seconds? Dave Cotton wrote: On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote: Derek Whitten wrote: [incoming] exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r) exten = s,2,Voicemail(myext) exten = s,3,Hangup() Thanks. This will call/ring multiple extensions but what about waiting for X rings before going to voicemail? How do I do that? What do you think the 25 does? Maybe it's a time or something. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Replication of a Single File
This sounds like a prime candidate for a database implementation. That way you can get very near real-time stats without the overhead of frequent cronjobs or polling. You number crunching computer would then just grab the data and crunch away. I'm just now getting started on using Asterisk in the more advanced modes (ie Realtime) so I do not know how to implement this, but I'm sure that it could be done. Ryan List users, Please provide me with tips on how to replicate a single file to a separate machine as changes are made to it. I would prefer a method that reacts to file modifications (ie. FAM/gamin) as opposed to timed loops/polling (cron + rsync). I'd also like to avoid NFS altogether. Keeping resource consumption low on the source machine is a priority. A bit of research has lead me to believe that calling rsync when gamin is alerted to a file modification would be a good fit for my scenario, but I'm unclear on the easiest implementation. My scenario is as follows. I have a machine that runs Asterisk VoIP PBX software. Asterisk creates a log file that we generate reports off of. Another machine handles the generation of these reports, which involves significant number crunching and file I/O. By replicating the file on the reporting machine, I'd like to decouple the resource consumption of reporting from the VoIP server. Some of the reports are used to monitor activities in realtime, so cronning off rsync on a large time interval is not an option. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I set up extensions.conf to dial out on analog telephone line?
Here is an example, very basic.
sip.conf
=
[1000];assuming that your first SIP extension is 1000
... ;all the other paramters
context=internal ;default context is [internal]
extensions.conf
[globals]
OUTBOUNDCHANNEL=Zap/2
[internal]
include = outbound
... ;rest of the [internal] context
[outbound]
exten = 911,1,Dial(${OUTBOUNDCHANNEL}/911}
exten = _9N.,1,Dial(${OUTBOUNDCHANNEL}/${EXTEN:1})
exten = _9N.,2,Congestion()
exten = _9N.,102,Congestion()
... ;the rest of the dialplans
This will allow anyone who dials a 9+number to call that number out Zap/2
(your FXO port). If it cannot dial out that port then the dialer heres a
busy signal. There is a specific line for 911 since it is so important.
Again, this is a very simple example, there have been more in depth ones
posted on the mailing list that when 911 was dialed checked if the FXO Zap
channel was in use and would hangup the channel then dial 911. not that
hard, but I don't remember how to do it off the top of my head (I'm still
learning :) ).
Hope this helps.
Ryan
I have one SIP phone (and soon a 2nd phone) and a Digium TDM11B (1 FXO +
1 FXS) card. I would like to be able to dial out the analog line via
Asterisk. How do I configure that?
i.e I'd like any extension to be able to dial 411, 911, 0, (617)
555-1212, 16175551212, etc... and have these routed out the POTS line.
Just like a regular telephone.
% cat /proc/zaptel/1
Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXSKS (In use)
3 WCTDM/0/2
4 WCTDM/0/3
% cat /etc/zaptel.conf
fxoks=1
fxsks=2
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Re: [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences!
While I'm not sure about the 1.2 ChangeLog the 1.2.1 that was released recently has an real Changelog that is linked at the top of Asterisk.org. http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.1 It looks like they put some effort into this one so I hope it continues for future releases. Ryan - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 06, 2005 11:31 PM Subject: RE: [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences! While we're on the topic, just what are those 3,000 improvements that Digium said where in Asterisk 1.2? Where are these improvements listed? Where are they documented? I'm sure Digum works insanely hard in the software and hardware department. I'd just like to see them work a little harder in the documentation deparment. Of course most of the problems are config problem, problems that could be fixed a lot faster if there was documentation. I think they'd also see a lot more sucess stories if Asterisk as a general rule addressed the issue of redundancy and the ability to add additional Asterisk servers for redundancy. No one wants to implement phone solution with a single point of failure. A good case in point is Asterisk realtime and it's apparent (I say apparent because it isn't officially documented anywhere) inability to share SIP contact information between Asterisk boxes in a common MySQL database. Call me crazy, but I would have thought this would have been a great way to share user contact info between multiple redundant Asterisk systems. -Original Message- From: Boris Bakchiev [mailto:[EMAIL PROTECTED] Sent: Tue 12/6/2005 10:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences! Why not? Digium works hard in hardware software department. It constantly improves its hardware offering. The software arm has been busier then ever! Million bug fixes, MANY MANY improvements, roadmap (at least from what I can see from contributing developers in SVN) is amazing. Asterisk and Digium have great feature together. Admittedly almost all had problems in one place or another but most of it is user/config problem. I would not have invested in Digium's hardware and taken up asterisk if I were not confident that Asterisk can cut it. If you take a look at general Digium Asterisk are success stories in itself! I'd volunteer for sure but my little installation probably a drop in the ocean compare to the ones I hear and read about and Australia is not exactly has competitive market for that. :) In fact I share the same view about any company that supports Asterisk community. Even for Digium competitors (who have the same dedication as Digium as well) Give it a chance, lets not forget that Digium spends great time, effort and expense getting Asterisk to where it is now. I don't know of many hardware manufacturers that do the same thing. Regards I was going to bite my tongue on my response to this, but keeping quiet is driving me nuts. If this is a legit post... In short, this irritates the heck out of me. Maybe if Digum supplied some documentation for less than $175/hr, then there might be a few success stories. The lack of any official documentation in my opinion is limiting the success of Asterisk. I seem to spend most of my Asterisk time researching people's personal heresay about how to get stuff to work. Often .the personal heresay is just someone else's heresay cut and pasted. Why the heck should anyone help Digium with good press in this instance? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and embedded system
On Mon, 21 Nov 2005 18:43:34 +0100, Josh wrote: Hi all, I'm kinda new with asterisk stuff. I'm running a Debian with asterisk and a digium X101P clone card in country #1. Since I'm going to work in another country (country #2), I would like to setup another Asterisk server + 1 FXO device in #2 as well as in #1. However I'm looking for a small solution. By small, I mean I don't wanna have a big desktop running 24/7 ... just an small box like a WRT54GS + ATA Ethernet FXO for ex ... I've read some howtos from http://www.voip-info.org/wiki/view/Asterisk+embedded+systems but I'm still wondering what kind of hardware to choose. The network I would like to setup is basic, something like : X101P + asterisk (#1) IAX asterisk + FXO (#2) Person in #1 will call the line connected to the X101P, then choose via a menu to call me. The call is transferred via IAX between the 2 asterisk boxes, then in #2, asterisk will dial the number via FXO With your experiences, which hardware/system (that can be reliable) do you recommend ? i was thinking of a WRT54GS + a GrandStream Handytone 488 (http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43151799552.htm), what do you think about this ? I heartily recommend Astlinux (www.astlinux.org) running on a Soekris Net4801. It boots in less than one minute to a CF card and stores all its setups, VM, MOH files, etc on a USB key drive. The whole setup will cost your around $220 USDmore than a WRT54G, but its far more flexible, reliable and supported by a great user community. It also has one PCI slot and one mini-PCI slot as well as an IDE interface on-board. Truly, I can't say enough good things about Astlinux. I've been using it for about 18 months. Prior to that I used Asterisk on Fedora Core 1 running on a VIA C5000 mini-itx, in an early attempt to make a fanless, silent system. Now for my next trickI'll again leverage Kristian's work (auther of Astlinux) to make an Asterisk installation running on a Gumstix acting as a SIP IAX2 protocol translator. That way ay SIP hard phone can be an IAX2 hard phone as well. That'd be my holiday season project. 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 fwd 54245 ___ --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 Micheal, Would you explain your setup with your Soekris Net4801 and AstLinux? I am curious as to how much the system actually gets used and possible any performance hints that you can give the list? I just looked at AstLinux and it is facinating... I'm just wondering how much a 266mhz 128 MB SDRAM system can handle. Might be good out of the box small business solutions that I might be able to recommend to my clients. Thanks, Ryan ___ --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] Is this maillist down?
Yep, I second (or third) that observation. Ryan It's not just him. The list was majorly down from sometime on the 29th until the 1st. MARK. Derek Whitten wrote: must be just you.. get messages all day every day here.. :-) On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote: This is usually a very active list, but looking at my procmail log the last message I have received arrived on: From [EMAIL PROTECTED] Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I don't think I have seen any mail from any of the asterisk lists, since that time so guessing this list is having some kind of problem... --- Howard Leadmon - [EMAIL PROTECTED] http://www.leadmon.net ___ 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] Is this possible?
Yes I would be very interested to see some kind of example if anyone has one running? Thanks - Original Message - From: Paul [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, April 01, 2005 11:19 AM Subject: [Asterisk-Users] Is this possible? I'd like to setup my Asterisk box to receive a call on the incoming POTS line and immediately redirect back out to connect to another phone number. Im thinking I could use either the threeway feature of that POTS line, or a second POTS connected to a different FXO card. Does ANYONE know if this is possible and if so, how it's accomplished? Paul ___ 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] Recommended Phone for beginner
Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like the Sipura SPA-2000. I've been looking at the Grandstream BudgetTone 100 series but after looking at the Wiki for setting up * with that phone it looks like it might be more trouble than its worth. Of course I would love a Cisco 79* but I'd like to keep the cost at a minimum but get a good amount of flexibility in tersm of features. Hopefully once I get over the learning hump I can start contributing to this list. Any input would be appreciated. Thanks, Ryan ___ 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] Recommended Phone for beginner
Wow.. what an awesome mailing-list! I appreciate the input, I'm looking at the Polycom 300 right now and debating on spending the extra $60 / phone to upgrade from the BudgeTone. Chris, I like hte Sipura 2100 idea, but I need at least 3 phones and would like to stick with native IP phones vs running my existing phoes through an adapter. Thanks for the suggestion though, it is definately good to know and a possibility in the future so that I can use some of my wireless phones with *. I'm always ready for more suggestions though.. Ryan - Original Message - From: Ryan Burke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 8:32 PM Subject: [Asterisk-Users] Recommended Phone for beginner Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like the Sipura SPA-2000. I've been looking at the Grandstream BudgetTone 100 series but after looking at the Wiki for setting up * with that phone it looks like it might be more trouble than its worth. Of course I would love a Cisco 79* but I'd like to keep the cost at a minimum but get a good amount of flexibility in tersm of features. Hopefully once I get over the learning hump I can start contributing to this list. Any input would be appreciated. Thanks, Ryan ___ 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] Recommended Phone for beginner
Great! That is what I was looking for and within my price range. Thanks. Ryan - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 9:32 PM Subject: Re: [Asterisk-Users] Recommended Phone for beginner Ryan Burke wrote: Wow.. what an awesome mailing-list! I appreciate the input, I'm looking at the Polycom 300 right now and debating on spending the extra $60 / phone to upgrade from the BudgeTone. Chris, I like hte Sipura 2100 idea, but I need at least 3 phones and would like to stick with native IP phones vs running my existing phoes through an adapter. Thanks for the suggestion though, it is definately good to know and a possibility in the future so that I can use some of my wireless phones with *. I'm always ready for more suggestions though.. Ryan Ryan, The Polycom IP300's can be difficult to configure - you either have XML config files, a small screen, or a limited web interface. Polycom's are my favorite phones, but I would reccomend a Sipura SPA-841 for this application. The lack of a built in switch could be a problem, but the 2.5mm headset jack helps ease the pain. The Sipura 841 goes for under $90. It works well, and has a nice web interface. Once you get more advanced you can use their Sipura Profile Compiler tools to work with multiple phones (or work through Polycom XML config files and switch to them)... -- Kristian Kielhofner ___ 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
