Re: [asterisk-users] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?
[EMAIL PROTECTED] wrote:
Hi all,
Does ENUMLOOKUP can query multiple DNS servers without having to
replicate the same code in which the only thing replaced is the server?
the enumlookup dialplan function (as opposed to the application) never
cares about your enum.conf file. The trick is to use separate enum
domains, and test them all in your dialplan using
${ENUMLOOKUP(+${ARG1:2:},sip,c,yourdomain.local) or something in a loop.
If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to
find the list of DNS servers in order of preference to be queried, but,
I pretend to use something like this: ${ENUMLOOKUP(+${ARG1:2:},sip,c)
which does not seam to care about the existence of enum.conf file! May I
force Asterisk to care about the servers I wrote in enum.conf?
To let you understand better, I wish to use just a block of code that is
able to query multiple DNS servers, instead of repeating like in the
following example the same code for each DNS server I wish to lookup for:
; Start first with e164.arpa zone:
exten = _X.,1,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,2,GotoIf($[${counter}${sipcount}]?3:6)
exten = _X.,3,Set(counter=$[${counter}+1])
exten = _X.,4,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,5,GotoIf($[${counter}${sipcount}]?3:6)
; ...then also try e164.org:
exten = _X.,6,Set(sipcount=${ENUMLOOKUP(+${EXTEN},sip,c)}|counter=0)
exten = _X.,7,GotoIf($[${counter}${sipcount}]?8:11)
exten = _X.,8,Set(counter=$[${counter}+1])
exten = _X.,9,Dial(SIP/${ENUMLOOKUP(+${EXTEN},sip,${counter})})
exten = _X.,10,GotoIf($[${counter}${sipcount}]?8:11)
; ...in case of no route by IP, then send out PRI:
exten = _X.,11,Dial(Zap/g1/${EXTEN})
Regards,
Ricardo.
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Re: [asterisk-users] Where an extension really is (DUNDi woes)
Kyle Sexton wrote: On 6/15/07, *Anthony Francis* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Kyle Sexton wrote: I have two servers setup to do DUNDi lookups against each other. The scenario is that on server A, I have a wildcard match for extensions 64XX that rings to a local extension on the server. On server B I have a 6442 real extension that I would like to have ring if called. It seems that DUNDi is matching on the 64XX and not searching out to see if there is a *more* exact match than the pattern match. Is there any way to get around this? I don't think I am incorrect in saying that dundi doesn't look for externally that which it knows about locally. I think thats pretty standard of routing protocols. I was afraid of that. It just means I have to explicitly list every number in the DID range (so hundreds of extensions). I was hoping DUNDi would make the dialplan simpler. :( it will, look into the regexten for your sip accounts. Now have dundi lookup a number in that extension... Now, you might need to abstract extension numbers from sip accounts, but that only has advantages anyway... -- Kyle Sexton ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Run as root?
Malcom Kemp wrote: In looking at the safe_asterisk script, it would appear that it is encouraging the running of the Asterisk application as root user. My natural inclination is to run it as a non-privileged user. What is recommendation? the recommendation is not to run as root, and have both the desired user and group configure in your asterisk.conf ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on capacity
Jerry Geis wrote: Can one server (like AMD 6000+ X2) with 2 GIG ram running asterisk 1.4 handle having 2100 wireless phones connected. All phones will not be talking at the same time only a couple will be. There may be 1 T1 card in the box. Will this work? If not how does one handle this situation. yes, it can work, but not with asterisk alone. SIP phones consume a lot of resources in asterisk. Better is a setup with a combo of (open)ser and asterisk, basically asterisk will be a gateway to the fixed phone net... Thanks, Jerry ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] My Kernel
bilal ghayyad wrote: Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I have the following: [EMAIL PROTECTED] /]# 'uname' -a Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1 SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386 GNU/Linux And when I type rpm -q kernel, then I have the followig: [EMAIL PROTECTED] /]# rpm - q kernel kernel-2.6.20-1.2319.fc5 So the question now is: what is my kernel that my system is using it? And how I can make my system use the latest updated kernel? Regards Bilal not to be rude, but what does this have to do with asterisk? From what you are telling us, I guess you need to find some fedora or general linux support medium... Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 ENUM entries and Asterisk fails to dial
[EMAIL PROTECTED] wrote: Hi, I use Asterisk 1.2.17 in my site, and now I'm trying to configure ENUM lookup in my server. When someone calls a number that has multiple ENUM entries, randomly Asterisk seems to fail to return a correct answer, and dial by ENUM fails. I've goggled a bit on this, but didn't get any good conclusion. There is some RFC Compliant ENUM Macro that can be used that is announced to solve this problem, but also it can be read that It seems that Asterisk 1.2.0 comes with a new powerful ENUMLOOKUP. So there is probably no need to use this script anymore.. Should I still use that macro? enumlookup works greak, as far as I can tell. It is already closed a Digium Issue Tracker concerning handle multiple records with the same order and priority, so, this problem shouldn't be arising anymore, shouldn't it? Does Asterisk 1.4 already solves this issue? just as well as enumlookup in 1.2 Regards, Ricardo. ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Scaling Asterisk: High volume benchmarks (0 to 450 calls)
594.001:1.5 8.98 91.02 55.77 36.624058022 400 --- - - - ---0 410 601.001:1.5 14.07 85.93 50.07 34.163798801 420 586.001:1.4 25.87 74.13 42.14 30.203759361 430 --- - - - ---0 440 --- - - - ---0 450 --- 4.33 95.67 50.82 -4134962 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Slow list
Philipp Kempgen wrote: Wow. My message made it to the list after more than 3 hours. Philipp I noticed similar delays, no wonder we get a lot of 'me too'-s to the list (sorry list for my bitching). -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] TCP-UDP SIP proxy?
Yehavi Bourvine +972-8-9489444 wrote: Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? (open)ser Thanks! __Yehavi: ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Voip-info.org
Compnet Bobby wrote: Same in southern cali! ok, it's down, we all know it, use the mirrors, stop spamming, thanks :) -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] WiFi SIP phones
Tony Plack wrote: Are the DECT phones two channel or do they share a channel like most other portable phones? DECT is a digital standard, quite distantly comparable to GSM. There are multiple channels (I believe the standard allows for 12 channels, but the last time I actually worked on DECT is ages ago). A siemens S450IP can have up two 6 handsets with 2 'external' (SIP or POTS) phonecalls concurently. You cannot decline a phonecall, but you can ignore it. The thing I like about the wired SIP phones is that they handle the echo issue fairly well. ATAs just reintroduce the echo issue of single pair type phones. DECT is completely digital, no echo ;-) I'd just like to say that I purchased a siemen S450IP recently and so far so good it's a nice handset and works better than previous wifi phones I've used. This is most likely due to it being dect gap where the base station handles the voip side and not the phone thus avoiding issues with 802.11 wireless and phone packets. Regards, Dee ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Working softphone for poket PC
Cosmin Prund wrote: Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). SJphone, and why did you remove it? Is there one (pocket pc softphone) that works? SJphone ;-) At least I've made some successful calls using sjphone Thanks, Cosmin Prund___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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: DUNDi configuration problem
Tim Verscheure wrote:
Does this even work?
exten = 5010,1,Dial(SIP/[EMAIL PROTECTED])
if priv is a sip account it does Yes, I guess you are on the right
track.
It keeps saying CHANUNAVAIL...
greetz
2007/5/22, Tim Verscheure [EMAIL PROTECTED]:
ok so now I changed ext-local to dundi-ext and I created this
context at the bottom of the extensions file. This is now the case.
[dundi-priv-canonical]
; Direct numbers
exten = 5010,1,NooP(DUNDI LOOKUP 5010)
exten = 5011,1,NooP(DUNDI LOOKUP 5011)
exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
exten = 5010,1,Dial(SIP/5010)
exten = 5011,1,Dial(SIP/5011)
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
[trydundi]
exten = _.,1,Macro(dundi-priv,${EXTEN})
exten = _.,2,Congestion
This is the dundi-ext at the bottom. In there I put this line:
[dundi-ext]
exten = _60XX,1,Dial(SIP/[EMAIL PROTECTED])
I made this myself, I think that if I get an incoming call from for
example 6010, the person would be dialing SIP/[EMAIL PROTECTED], right?
this is the output:
*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/6010-0820cdc8,
dundi-ext|5011|1) in new stack
-- Goto (dundi-ext,5011,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/6010-0820cdc8,
SIP/[EMAIL PROTECTED]) in new stack
[May 22 14:36:18] WARNING[4520]: chan_sip.c:2738 create_addr: No such
host: priv
[May 22 14:36:18] WARNING[4520]: app_dial.c:1099 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/6010-0820cdc8' status is
'CHANUNAVAIL'
2007/5/21, Remco Post [EMAIL PROTECTED]:
Tim Verscheure wrote:
Now I get this... If I call from 5011 on the 192.168.1.103 machine to
6010 on the 192.168.1.69 machine my X-lite softphone says, call
declined
this is the output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508,
ext-local|6010|1) in new stack
-- Goto (ext-local,6010,1)
[May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel
'SIP/5011-081da508' sent into invalid extension '6010' in context
'ext-local', but no invalid handler
so, is there an extension 6010 in you context ext-local? Probably
not ;-)
I'll add my extension file so you can see it. greetz
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High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167
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computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end. -- Douglas Adams
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Re: [asterisk-users] Help installing on OpenSuSE 10.2
Malcom Kemp wrote: make[1]: g++: Command not found hint :) -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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: DUNDi configuration problem
Tim Verscheure wrote: Now I get this... If I call from 5011 on the 192.168.1.103 machine to 6010 on the 192.168.1.69 machine my X-lite softphone says, call declined this is the output: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508, ext-local|6010|1) in new stack -- Goto (ext-local,6010,1) [May 21 15:32:46] WARNING[8939]: pbx.c:2450 __ast_pbx_run: Channel 'SIP/5011-081da508' sent into invalid extension '6010' in context 'ext-local', but no invalid handler so, is there an extension 6010 in you context ext-local? Probably not ;-) I'll add my extension file so you can see it. greetz 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: If I read all this is realize what a noob I am in this matter. Could I make a call by saying something like this: exten = 16000,1,Dial(SIP/[EMAIL PROTECTED]) you could, look into the DUNDILOOKUP function... Or something like that? 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: like this??? [dundi-priv-switch] ; Just a wrapper for the switch switch = DUNDi/priv yes that should do. Does your asterisk console show anything useful? And if you do wind up in the switch, what does you dundi debug show? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Dealing with 2 SIP providers
Brad Templeton wrote: On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote: Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available there is a macro floating around called safedial. Basically this does what you want, if one provider is unresponsive, use the other. You could possibly have 2 variations, so you can first try your preferred provider for that extension and then fail-over to the other. My provider provides 2 uplinks, so I randomly select one of the two... What would be really cool, but require special code in the chan_sip dialer, would be automatic support of multiple providers in a similar fashion to the way Asterisk can ring two channels and only talk to the first to answer. You can't just do this with outgoing providers, because if you try to ring two at once, you may very well have the second one go to a voicemail and thus answer right away (because the first is ringing) and you would treat that as the success. What I have in mind is something like this: a) Invite to main provider b) Await some intermediate response, such as a RINGING code or some early media c) If you don't get that after a short timeout (more like 5 seconds) then INVITE the second provider d) Upon the receipt of a ringing or early media code from either, CANCEL the other. Now you would have to get your timings right because there could still be risk of doing something bad, such as a 2nd call going to voice mail or residual ringing making a call waiting on the recipient. (I don't know what typical 5ess do with a 2nd call that comes in while still ringing, anybody known?) Anyway, this could be a good course when a provider has known unreliability. Long timeouts and restarts are very annoying to 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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Ser vs. DUNDi
Curt Shaffer wrote: With all of the recent talk on the list about DUNDi, I have a question. From the outset it appears that SER is often used for high availability solutions and as a tool for almost clustering Asterisk boxes behind it. It appears to me that DUNDi is providing a lot of this as well. Now I know DUNDi is not an application by itself to proxy SIP requests but can I hear any information out there that supports that DUNDi is in fact a valid alternative to something like SER or not? A nice feature analysis between the two in a clustering/highly available solution would be nice to see. Not a feature list but rather a discussion from people that have tested/used both for people who are deciding which way to go to achieve the goal. I'm currently 'using' dundi and it works great for small environments. I guess that up to several tens of nodes and a few concurrent calls going on, a set of * boxes clustered using dundi is quite an elegant solution. Now, if you have hundreds or more clients all registering quite frequently, you * boxes will have an hard time keeping up. In that case, it makes lots of sense to set up a set of (Open)SER registrars to handle that load and have * take on the role of a 'border' gateway, of course still clustered to achieve maximum availability and an opportunity to take one down for maintenance without impacting production too much. Now, I haven't had time to test this, but from what I hear, this is the way o go for larger environments. I haven't had time to experiment with this, but it seems feasable. Thanks! Curt ___ --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 -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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: DUNDi configuration problem
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication happening? Test
with 'dundi lookup number bypass'.
greetz
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial
Your mappings are wrong, this is for IAX, for SIP to work, it should be:
priv = dundi-priv-canonical,0,SIP,${NUMBER}@the real IP
Address,nopartial
The rest looked ok I think.
Good luck.
JR
--
JR Richardson
Engineering for the Masses
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SARA - Reken- en Netwerkdiensten http://www.sara.nl
High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC
I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end. -- Douglas Adams
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Re: [asterisk-users] Asterisk on OpenSuSE 10.2
Malcom Kemp wrote: I am new at this. I have read Asterisk: The Future of Telephony and have installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5). The GUI interface does not seem to provide the capability that I need, although I have modified the *.conf files to successfully create what I need. Given this, I would like to install Asterisk on a distro. I am most familiar with SuSE, and have the OpenSuSE 10.2 distro. I downloaded asterisk-1.4.4.tar.gz, zaptel-1.4.2.1.tar.gz, and libpri-1.4.0.tar.gz. I have untarred these, and have tried to run make. When I run make clean on zaptel, it give me an error about an included makefile not having a clean target. When I attempt to run make on Asterisk, the make clean runs, but when I try make, it gives me a message saying I have to run configuration script first. If I run make config, it give me a message that the distribution is not supported. for asterisk, try ./configure in the source dir (yes, this should be in the README), then run make and as root make install. zaptel can be build by just running make and make install. Has anyone put Asterisk on the 10.2 distro? Any pointers? Thanks. + This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit www.tecinfo.net + ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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: DUNDi configuration problem
Remco Post wrote:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication happening? Test
with 'dundi lookup number bypass'.
'dundi lookup number@priv bypass' of course
greetz
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial
Your mappings are wrong, this is for IAX, for SIP to work, it should be:
priv = dundi-priv-canonical,0,SIP,${NUMBER}@the real IP
Address,nopartial
The rest looked ok I think.
Good luck.
JR
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PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC
I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end. -- Douglas Adams
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Re: [asterisk-users] Re: DUNDi configuration problem
Tim Verscheure wrote:
I tried to call to with X-Lite to extension 6000 but it still doesn't
go through:
Call Failed: Not found
so you'll have to set de dundi context in your switch statement as well,
else the switch will look in an e164 context, that doesn't exist.
greetz
2007/5/19, Remco Post [EMAIL PROTECTED]:
Remco Post wrote:
Tim Verscheure wrote:
still nothing... it gives DUNDi lookup returned no results. DUNDi
lookup completed in 0 ms
have you set dundi debug? Is there any communication happening? Test
with 'dundi lookup number bypass'.
'dundi lookup number@priv bypass' of course
greetz
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-customers,100,SIP,${IPADDR}/${NUMBER},nopartial
priv = dundi-priv-via-pstn,400,SIP,${IPADDR}/${NUMBER},nopartial
Your mappings are wrong, this is for IAX, for SIP to work, it
should be:
priv = dundi-priv-canonical,0,SIP,${NUMBER}@the real IP
Address,nopartial
The rest looked ok I think.
Good luck.
JR
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Met vriendelijke groeten,
Remco Post
SARA - Reken- en Netwerkdiensten http://www.sara.nl
High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC
I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end. -- Douglas Adams
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Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] Re: DUNDi configuration problem
Tim Verscheure wrote: like this??? [dundi-priv-switch] ; Just a wrapper for the switch switch = DUNDi/priv yes that should do. Does your asterisk console show anything useful? And if you do wind up in the switch, what does you dundi debug show? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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: DUNDi configuration problem
Tim Verscheure wrote: If I read all this is realize what a noob I am in this matter. Could I make a call by saying something like this: exten = 16000,1,Dial(SIP/[EMAIL PROTECTED]) you could, look into the DUNDILOOKUP function... Or something like that? 2007/5/19, Remco Post [EMAIL PROTECTED]: Tim Verscheure wrote: like this??? [dundi-priv-switch] ; Just a wrapper for the switch switch = DUNDi/priv yes that should do. Does your asterisk console show anything useful? And if you do wind up in the switch, what does you dundi debug show? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Anyone tested the new Sony Ericsson P1 phones..
Rosli Sukri wrote: Hi, Has anyone on this list tested out the new SE P1 phones (http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says it supports VOIP, wonder if it is working with asterisk. Nice phone, wondering what it will cost when it gets released. So to answer your question, not very likely that someone has, since it's not available yet -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] The purpose of DUNDi
dave cantera wrote: remco, et al, could I use dundi where I could use an area code to determine the connecting server or dial string? just like we would use 88XXX to dial a 3 digit extension on another server at location 88? or dial 84XXX for a 3 digit extension on a server located at 84?... yes you can. You'll setup a context in your dialplan on your server where you'll tell dundi that you accept calls for say _88XXX and have a mapping for that context in your dundi.conf thanks, daveC Remco Post wrote: Rilawich Ango wrote: It is quite interesting and I am looking for it. Could you give me some more information or website how to set it up? Have a look at: http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf and the two links at: http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] The purpose of DUNDi
Rilawich Ango wrote: How about if both ServerA and ServerB houses extensions 500 throught 699. Such that users can dynamically register Server A or Server B. Can we use DUNDi to implement such network? yes. Use regexten and regcontext and have dundi look into that context to see if an extension is available. The actual call doesn't need to go trough that context. This is the way to go for HA clusters. On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote: Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf ). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] The purpose of DUNDi
Rilawich Ango wrote: It is quite interesting and I am looking for it. Could you give me some more information or website how to set it up? Have a look at: http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf and the two links at: http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk 1.4 depoyment.
Vietnhi Phuvan wrote: Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can I either fix or work around this? Is Asterisk 1.4.2 ready for production deployment? asterisk 1.4.2 is certainly not, 1.4.4 otoh is. I don't have any experience with meetme. Do you have zaptel loaded? Meetme depends on zaptel (at least ztdummy) for timing. Regards, Vietnhi -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] MYSQL Query -- PAGE
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from sip where extension like '6%'
6001
6002
6003
ex
I need to put all the results into a variable that would equal something
like:
SIP/6001SIP/6002SIP/6003
I have setup a couple basic MYSQL Query's for my dialplan. Mostly
just looking up a DID to Extension Mapping for setting callerid on
outbound and inbound calls.
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Something like Set(devices=${devices}${newrow_result})
I looked at the example on
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL but that doesn't
seem to be accurate.
Thanks all!!
What I've done in postgresql is to build an pl/pgsql procedure that
returns the desired dialstring. So the procedure does the select and
then concats them.
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Mark Coccimiglio wrote: Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. at least on ubuntu 6.10 safe_asterisk requires one simple fix, not really a headbreaker (something with output redirection). You could actually edit the script to not start a console if you dont' want it to (say for security reasons). If you wanted to start asterisk and keep monitoring it, that is what init is for. I don't know about ubuntu startup, but traditional sysV init would simply restart a process if it ever quits (respawn). My bet is that startup can do the same somehow, this is a far better way to keep * up -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Starting Asterisk on Ubuntu 7.04
Tzafrir Cohen wrote: On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote: Mark Coccimiglio wrote: Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is THE TOP priority. If you would rather check every five minutes then replace the first * with */5. I will address your points as it seems that you haven't really thought about this. 1) In a production environment you should NOT be messing with the config. That's what test hardware is for. 2) The answer to this question is: crontab -e its really not that hard. I'm not running asterisk every minute. I'm looking to see if asterisk is running and then act accordingly 3) If asterisk fails believe me a full mailbox is the least of my worries. As for full logs I'd rather have more informationgrep awk are your friends. I prefer to keep things as simple as possible. Sure scripts like safe_asterisk are nice and do some really neat things but lets face it how often do you actually sit at the console of your asterisk box. My main PBX is located about 7 feet from my office desk and I still mostly use ssh (not even telnet) to get into the box. at least on ubuntu 6.10 safe_asterisk requires one simple fix, not really a headbreaker (something with output redirection). Bashism? The rule in Debian is that a bourne shell script (#!/bin/sh) should not use bash-specific features, such as . If it does, it should explicitly ask for bash: '#!/bin/bash' hmmm, you might have a point there, never thought of that. You could actually edit the script to not start a console if you dont' want it to (say for security reasons). Could you please elaborate? Change: CONSOLE=yes # Whether or not you want a console To 'CONSOLE=no' I believe that this would wreck the error handling in that script. If you wanted to start asterisk and keep monitoring it, that is what init is for. I don't know about ubuntu startup, but traditional sysV init would simply restart a process if it ever quits (respawn). My bet is that startup can do the same somehow, this is a far better way to keep * up But this means editing /etc/inittab every time you actually want to stop asterisk. Or change runlevel... well that is maybe a bit to much AIX :) -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] I'm looking for solution
Ardit Saliu wrote: HI I have 3 Linksys SIP901 IP phones I also have a pc I’m not using it amd athlon 1800+ 512mb ram and 40 gb hdd I’m looking to connect this phones together and to make calls between them Not from outside of my lan I don’t know how to configure asterisknow beta Can somebody help I’m doing this in my house to connect rooms Have you looked at http://www.asterisknow.org/files/downloads/quickstart_asterisknow.pdf ? With respect Ardit Saliu ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Connections rejected in DUNDi requests
Chris Bagnall wrote:
Greetings list,
Wondering if anyone's come across this before.
I've configured a couple of our servers with a privatedundi context to
allow calls to still flow between extensions even if they're registered to
different servers . The DUNDi lookups seem to work fine, evidenced by the
following on the originating server:
-- Called private:password@ip/[EMAIL PROTECTED]
shouldn't that be 'private:password@ip/minotaur-201'? I guess you
have a mistake in your dundi mapping
However, on the destination server, I have the following:
May 4 03:50:45 NOTICE[1149]: chan_iax2.c:7354 socket_read: Rejected connect
attempt from 80.68.80.210, request '[EMAIL PROTECTED]' does not exist
I then performed the following:
cronus*CLI show dialplan privatedundi
[ Context 'privatedundi' created by 'pbx_config' ]
'_minotaur-2XX' = 1. NoOp(Connected to ${EXTEN})
[pbx_config]
2. Goto(minotaur|${EXTEN:9}|1)[pbx_config]
Unless I'm missing something, [EMAIL PROTECTED] definitely *does* exist.
I've tried manually specifying minotaur-201 in full rather than as a pattern
match - which works correctly. I'm having exactly the same problem the other
way around (origination and target servers reversed).
What's particularly strange is that other entries in [privatedundi] such as
_clienta-2XX, _clientb-2XX are working fine between the same servers.
So, what's special about _minotaur-2XX vs. _somethingelse-2XX that causes
pattern matching to fail?
If anyone can shed some light on this I'd be most grateful.
Regards,
Chris
--
Met vriendelijke groeten,
Remco Post
SARA - Reken- en Netwerkdiensten http://www.sara.nl
High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167
PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC
I really didn't foresee the Internet. But then, neither did the
computer industry. Not that that tells us very much of course - the
computer industry didn't even foresee that the century was going to
end. -- Douglas Adams
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Re: [asterisk-users] SIP RealTime Friends
Forrest Beck wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. have you tried? If so, what went wrong? (*hint* ;-) ) -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] OT: USB T1/E1 Interface?
Michael Collins wrote: Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I’ve seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Thanks! -MC ___ --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 -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] SIP Proxy
Ronaldo wrote: Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router http://www.voip-info.org/wiki/view/SIP+Express+Router: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org http://www.voip-info.org/wiki/view/Vovida.org * sipX http://www.voip-info.org/wiki/view/sipX from Sipfoundry http://www.voip-info.org/wiki/view/SIPfoundry is a native SIP proxy but also a complete SIP PBX * OpenSER http://www.voip-info.org/wiki/view/OpenSER - scalable and robust SIP server with TLS support Can anyone suggest me something about these SIP Proxy? SER and OpenSER are related. SER is more geared towards stability, OpenSER towards features. I guess that If SER does what you need, go with SER, otherwise look at OpenSER. I think that for instance freeworddialup uses SER. I'd think it is a very viable solution. The other two I don't knnow. p.s) Is Asterisk a SIP Proxy? No, it is not, * is a back2back user agent. Regards ... Ronaldo. ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 1.4.4 VoiceMail ODBC Storage Help
Bruce McAlister wrote:
Hi Remco Post,
Having read your patch I suggest you report this bug at bugs.digium.com,
it seems to be legit...
However, I think I may have got to the bottom of the issue now.
What look like was happening is that asterisk was trying to delete any
matching row prior to an insert operation. So, when a user left a message,
for example, message 1, asterisk would attempt to delete message 1 before
inserting it for that user. However, message 1 does not exist at that time
and thus the ODBC driver returns SQL_NO DATA.
The same happens when a user checks their voicemail, once an message has
been listened to asterisk moves it to the Old directory, that way it can
distinguish between new/old messages. When a user listens to the voicemail,
asterisk then tries to insert the message into the Old tree, prior to
doing the insert, asterisk tries to delete the last available message
returned from a select count(*) operation. This message does not exist and
the odbc driver returns SQL_NO_DATA.
The delete_file function in app_voicemail.c does not accommodate for this
return code SQL_NO_DATA and thus spits out the warning on the console.
I thus changed the following condition in function delete_file in
app_voicemail.c from:
if ((res != SQL_SUCCESS) (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, SQL Execute error!\n[%s]\n\n, sql);
SQLFreeHandle (SQL_HANDLE_STMT, stmt);
ast_odbc_release_obj(obj);
goto yuck;
}
To:
if ((res != SQL_SUCCESS) (res != SQL_SUCCESS_WITH_INFO) (res !=
SQL_NO_DATA)) {
ast_log(LOG_WARNING, SQL Execute error!\n[%s]\n\n, sql);
SQLFreeHandle (SQL_HANDLE_STMT, stmt);
ast_odbc_release_obj(obj);
goto yuck;
}
This seems to have fixed the problem.
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] allowing call to my pabx every 15 minutes
Goke Aruna wrote: Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. Won't IfTime do the trick? I will be glad, if someone can give me a hint on this. Goksie ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] ADSL routers with integrated SIP QoS for other devices
Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which I'll never visit (minimum 3 boxes - ADSL modem, m0n0, WiFi AP). So, does anyone have any recommendations for a wireless ADSL router with integrated QoS for SIP/RTP? I've looked at some of the Draytek units (e.g. Vigor 2700V), but I can't find reference as to whether the integrated QoS applies only to the FXS ports in the router itself, or to all SIP traffic (most of the users will have separate SIP hardphones). These are all to be used in the UK, so the device in question needs to support PPPoA. Any suggestions gratefully appreciated. There are quite a few software stacks for the linksys wrt54g routers/ap's, some of them supportq qos. I hear good things about the wrt routers in general. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 1.4.4 VoiceMail ODBC Storage Help
Bruce McAlister wrote: Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: It looks like the * database user doesn't have permission to delete records from the voicemailmessages table. Make sure it has the propper permissions on that table. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] MYSQL application in dial plan
Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? or, you can use func_odbc that comes with * 1.4. Now you don't have to connect to your db every time you use it, I think. Thanks, __Yehavi: ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 dundi worth pursuing in this situation?
Salvatore Giudice wrote: DUndi or enum only make sense if you plan to move extentions dynamically without having to touch you Asterisk configs or if you want to expose your addressing to the outside world. Personally, I would do it statically so you can avoid delays in processing addressing especially - in the case of enum- if you dns server becomes unavailable. I've been using both enum and dundi. Dundi has some means of setting one server to primary, so that server will most likely have the number you are looking for in it's cache. With enum, you'll want to run a dns recursor on each * host or on a host very close to it networkwise. Both will do equally well in your case. I like the easy of using dundi. There are some very good dialplans floating around for doing enum lookups, I have a macro written in ael2 that you can have if you like. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Delay in Dial()
Eric ManxPower Wieling wrote: Stephen Bosch wrote: Eric ManxPower Wieling wrote: Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. [extensions] exten = 666,1,Dial(Local/[EMAIL PROTECTED]@Local/[EMAIL PROTECTED]) exten = desk,1,Dial(SIP/deadbeef-a) exten = cell,1,Wait(15) exten = cell,2,Dial(Zap/G1/5551212) Wouldn't just using the Dial timeout option do the same thing more elegantly? Or do you want the SIP phone to keep ringing? No. Dial(SIP/deskSIP/cell) would find BOTH phones at the same time. The original poster wants the desk phone to ring, then after X seconds, KEEP ringing the desk phone, but also ring the cell phone. Using the Dial timeout would STOP ringing the desk phone then, start ringing the desk phone again and also ring the cell phone. and how long in seconds would you think it takes * to step from the first dial to the second? Is this a real risk? I don't know about you but it would seem pretty unprofessional to me if my deskphone rang, I went to pick it up, got a dialtone because I did not get to it in time, then before I hungup the deskphone Asterisk rang both the desk and cell phone. Since the deskphone is offhook the call could go immediately to voicemail and then there would be no call when you rushed over to pick up the cell phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Applet?
Pablo L. Arturi wrote: Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I have tried JIAXClient, but it allows people to call anywhere, and what I want is just a configurable applet for letting people call me directly with a single click. so, if jiaxclient is ok, just drop those clients in a context that allows them to call just you Anyone? Not sure if this question is off-topic, if so, please accept my apologizes. Thank you, Pablo ___ --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 -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] don't want call to get answered
Arun Kumar wrote:
In my * box I've configured two queues and incoming number and whenever
any one calls those number call comes to my *box and it sends call to my
agents in queue. but if no agent is available it still answer the call.
Is there any why when my agents are not available I don't want call to
get answered. Here is my dialplan:
I think that you want asterisk not to pick up the call when all of your
agents are busy, right? I guess that in this case you do not want to use
queues, but you want to build some dialgroup.
So you'll need to do a few things
1- have some extension that agents can call to log on. When they do,
append their account (Technology/resource) to a database record (or
global var).
2- have some extension that agents can call to log off. Reverse of the
above.
3- When somebody calls the extension '' from below, you use Dial on
the database entry or global var to call all of your agents. If the dail
fails, you check the dialstatus to see why and possibly retry after so
many seconds (for a limited amount of tries) and then maybe answer the
call to play an announcement that nobody is available and please try
again later, or would they want to wait an be placed in a queue.
exten = ,1,GotoIfTime(*|*|20|dec?ccagents,,6)
exten = ,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,,7)
exten = ,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,,7)
exten = ,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,,7)
exten = ,5,GotoIfTime(09:00-18:00|mon-fri,*,*?ccagents,,7)
exten = ,6,Goto(out-of-hours,5003,1)
exten = ,7,Answer()
exten = ,8,Playback(custom/next-avail-advisor)
exten =
,9,Set(MONITOR_FILENAME=/var/spool/asterisk/q/talksupport-${TIMESTAMP}-${UNIQUEID})
exten = ,10,Monitor(wav,${MONITOR_FILENAME},mb)
exten = ,11,Queue(kbsupport,t)
exten = ,12,Hangup()
thanks
arun
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--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] Send Variable in Dial
Andres Gomez wrote: Hello to all I need send a data to sofphones screen when I use a Dial () . There is the applications SendText, SendImage or SendURL. Also, for SIP phones you could possibly use SipAddHeader... Thanks a lot Regards Andres Gomez ___ --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 -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Improving Asterisk's DNS support
Kristian Kielhofner wrote: Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with iffy DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) wasn't there some setting for that? I run a dns caching deamon om my * box (speeds up enum lookups big time), but i seem to recall that some dns settings could be made - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS problems - why? because your * can't resolve the names any more? - dnsmgr doesn't support SIP (yikes!): http://bugs.digium.com/view.php?id=9153 - other randomness (please contribute your own experiences) What can we do about improving this situation? At the very least we need to extend DNS manager support to SIP. I'm willing to pay for this and any other Asterisk DNS improvements. Any other ideas? -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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] Voicemail on Different Server
Eric Germann wrote: How do you handle transfering vmail from one user to another when they're on separate servers? I'd have a look at: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage Having voicemail stored in a database solves all kinds of potential locking problems. I guess (never played with it) that voicemail messages are accessable from all * servers that have access to that table. I'm using the single vmail server, mounted NFS partition for this right now. I'd love to be able to have them standalone so they're survivable when the WAN collapses, but I haven't figured out transfer. EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Saturday, April 28, 2007 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail on Different Server Hi Steve - Can you elaborate on this, I changed to storing the voicemail via ODBC on MySQL. Each server had it's own local storage, and then MySQL replicated the databases between the sites. This setup was terribly finicky and unstable. It was much worse than the NFS mount. I quickly gave it up. This sounds like it would probably work the best, especially if you have users moving around between offices. What was so finicky and unstable about it? I am not one to quickly give up. I have found that persistence pays off when the idea is sound. Yeah, I thought I had found the silver bullet with MySQL replication (the users do float between offices, so it seemed perfect). There were a number of problems, but in the end it was table corruption as a result of the replication process that made me drop this solution. At the time I set this up, MySQL replication was really designed for one-way replication. Two way replication was possible, but required somewhat unorthodox methods. (Maybe this has changed, I don't know). Configuration is also a little tricky. It's not too bad to set it up between two machines, but 3 machines is more tricky, and 4 is even more tricky, etc, etc. This client had only 3 offices at the time, but I knew they would be expanding. They now have 6. Anyway, after getting everything working, I found that replication would periodically stop after some time. I'd have to re-create the setup, and then replication would work for a time, and then stop again later. This occurred across several different version of MySQL. I suppose I could have fixed this issue with persistence, but unfortunately this was only an annoyance compared to the major issue of data corruption. When replication worked, it was inevitable that after a time the voicemail storage table would experience data corruption. Asterisk did not handle this gracefully at all. It was effectively a total DOS. This also occurred across several versions of MySQL. Sometimes I was able to repair the tables, but usually I couldn't, and the users ended up losing quit a lot of voicemails. I did not have the ability to spend the amount of time I needed to fix the issue, so I scrapped the whole setup. Regular local voicemail storage has been flawless in all installations I've administered. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 Pix firewalls
Lee Jenkins wrote: Is it possible to reduce the number of ports to be opened if there is moderate traffic? YEs, you could set rtpstart and rtpend in rtp.conf to whatever. I Have rtpstart 1 rtpend 10100 This is about enough for 25 concurrent conversations -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] dundi problem * 1.4.2
Asterisk [Submusic] wrote: entityid=00:00:F8:04:C4:51 ok, thanks, found it. The setting is entityid, not entity... easy to mis I guess ;-) Maybe some more verbose parsing would have helped ;-) -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] dundi problem * 1.4.2
Hi All,
I've been banging my head on a small dundi problem...
I have two * servers setup, both have almost identical dundi.conf files:
[EMAIL PROTECTED]:/opt/asterisk/etc# cat dundi.conf
[general]
department=thuis
organization=pipsworld
locality=Amsterdam
stateprov=NH
country=NL
[EMAIL PROTECTED]
phone=+31207508308
;bindaddr=0.0.0.0
;port=4520
entity=00:02:b3:49:69:5e
ttl=16
autokill=yes
;secretpath=dundi
[mappings]
;pipsworld =
pipsworld,1,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
;pipsworld =
external,1000,IAX2,[EMAIL
PROTECTED]/31207508308,nounsolicited,nocomunsolicit,nopartial
[02:60:8c:f2:3e:aa]
model = symmetric
host = pipc.pipsworld.nl
inkey = pipsworld
outkey = pipsworld
include = pipsworld
permit = pipsworld
qualify = yes
and:
[general]
department=thuis
organization=pipsworld
locality=Amsterdam
stateprov=NH
country=NL
[EMAIL PROTECTED]
phone=+31207508308
;bindaddr=0.0.0.0
;port=4520
entity=02:60:8c:f2:3e:aa
ttl=16
autokill=yes
;secretpath=dundi
[mappings]
pipsworld = pipsworld,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
; pipsworld =
external,0,IAX2,[EMAIL
PROTECTED]/${NUMBER},nounsolicited,nocomunsolicit,nopartial
[00:02:b3:49:69:5e]
model = symmetric
host = tsjonge.pipsworld.nl
inkey = pipsworld
outkey = pipsworld
include = pipsworld
permit = pipsworld
qualify = yes
But for some reason dundi-lookups fail.
tsjonge*CLI dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 3 ms
ETx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: DPDISCOVER
(Command)
Flags: 00 STrans: 23682 DTrans: 0 [145.100.55.14:4520]
VERSION : 1
DIRECT EID : 00:50:da:73:18:c6
CALLED NUMBER : 29
CALLED CONTEXT : pipsworld
TTL : 16
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 23682 DTrans: 0 [145.100.55.14:4520]
ENTITY IDENT: 00:50:da:73:18:c6
KEYCRC32: 1754443205
ENCDATA : [IV a7a14a4b63ccc33286b2b23108b05f49] 4 encrypted
blocks
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 21677 DTrans: 23682 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 23682 DTrans: 21677 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 15333 DTrans: 0 [145.100.55.14:4520]
ENTITY IDENT: 00:50:da:73:18:c6
SHAREDKEY : [ 5b c1 3c b5 41 6d a9 11 62 40 16 0a a4 b9 11 1f
54 ae b1 7f bd af de f7 aa 5a 72 13 2e d8 b1 e7 56 17 4a 48 6a 82 3b 66
ef c4 07 b7 ce 3e ab 39 d0 75 b4 b4 0f 08 af 21 9f d6 a9 45 34 be bd 59
bc e2 a2 5b a3 d8 60 7d 8d d2 31 01 24 73 ba 27 e0 3d ce ca 22 50 c6 ef
83 ba b6 24 b3 7d 34 5b c2 c0 31 36 b5 1d bf 62 73 56 77 61 b5 5f 9e cf
d3 d2 8b 98 25 e6 47 54 7f a6 0f 97 42 ab 96 74 ]
SIGNATURE : [ d3 d9 4f d2 05 9d 71 b3 4f 76 32 29 74 02 51 2f
90 40 10 c8 6c 49 3d 67 e4 8b e4 bd 2b ca 32 ed 65 d3 b0 bc 87 ff 30 60
05 e6 f2 e2 52 2f 04 6a a4 6a fe 6e ca 9c d0 e5 24 fa e6 35 9d 38 0a 93
61 46 84 04 03 c2 f8 9d eb b5 06 60 5b 23 f3 33 69 82 3c ba 2c 57 f9 af
1a be a9 b5 23 0d 53 58 f0 fa 07 13 c1 79 b8 37 5e 7c 87 dc 14 1b a3 ec
78 6e 91 8d 1d fa 52 db 54 ce 03 3e d8 ac 96 86 ]
ENCDATA : [IV a7a14a4b63ccc33286b2b23108b05f49] 4 encrypted
blocks
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 15402 DTrans: 15333 [145.100.55.14:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
as you can see from the dialplan the extension is available:
pipc*CLI dialplan show pipsworld
[ Context 'pipsworld' created by 'IAX2' ]
'20' = 1. Noop(remco)[IAX2]
'22' = 1. Noop(tsja) [IAX2]
'23' = 1. Noop(sipura1_tst) [SIP]
'24' = 1. Noop(sipura2_tst) [SIP]
'28' = 1. Noop(s450_1) [SIP]
'29' = 1. Noop(s450_2) [SIP]
'sipura1_lijn' = 1. Noop(sipura1_lijn) [SIP]
'sipura2_lijn' = 1. Noop(sipura2_lijn) [SIP]
also, tcpdump shows that both dundi-peers are communicating (as does the
dundi debug output).
Any hints?
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Re: [asterisk-users] dundi problem * 1.4.2
Asterisk [Submusic] wrote: Hi, I'm not working with Asterisk 1.4.X, but i think your Dundi.conf is not correct. well, things haven't changed in the dundi.conf going from 1.2 to 1.4, so that should be ok. If you want i can send you my complete working exemple with Asterisk 1.2.x (I think the config is the same) Please do. I've had a friend look at my dundi.conf, he couldn't find anything wrong with it, but it is quite likely that there is. Fred -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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 M$ SQL Server
Callum McGillivray wrote: I was hoping for something more along the lines of the Asterisk CMD MySQL(). I could always resort to something like that.. but I don't want to run it on a windows server and I really don't want to go to the bother of writing FastAGI scripts to make it all happen. I just want to write a quick and dirty DB lookup so that I can rewrite the CLI before passing the call along. IN Asterisk 1.4 you could use func_odbc. I guess this will work as well with mssql as with postgresql, which is quite satisfactory. Also, with cdr_odbc you could store cdr records in you database. mitcheloc wrote: Oh. Got it now. Well, in this case I think you are looking at it backwards. I imagine most users with this requirement write AGI scripts that talk to their databases then communicate back. You can use FastAGI and run your code on a windows server, or you could use any other programming language (PHP, MONO) to acheive the same, but then place it locally on the Asterisk side. On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Oh Microsoft SQL Server for those unfamiliar with the term M$ ;) mitcheloc wrote: I've never heard of M$ SQL Server? On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could point me to a howto / guide or relate their experiences with this, that would be great ! Thanking you all in advance. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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: asterisk-users Digest, Vol 33, Issue 80
Robinson C P wrote: Hi all, * i am using widows based asterisk pbx(AstWin) which i have down loaded from www.asteriskwin32.com http://www.asteriskwin32.com . we r using x-lite as a soft phone now .we have created 10 sip users in sip.conf and configured extensions.conf too. all of us could make calls through asterisk. we made 10 calls at the same time through one peer. now i wanted to transfer the call from one user to another. how can i do that with out sip signals. i mean, with dial plan. with out x-lite? * do asterisk can communicate with sqlserver-2005...? I don't know which asterisk version you are running. In version 1.2 there is at least the possibility to store cdr records in an odbc database, I'm using version 1.4.2 to do far more intresting things in my dialplan using func_odbc to route calls to groups of users and maintain an on-line telephone directory * how can we create a predictive dialer...? do asterisk can make calls itself..? what is the basic concept behind it..? I'm not sure what you mean. Asterisk can initiate a call, just drop a callfile in the queue, this is pretty wel documented on www.voip-info.org * do asterisk can make call itself or do we have to send sip signals which is saying that please dial this number...? * can we do the recording part with the help of asterisk( i mean, using monitor() application)...? yes you can. * how we can work on meet me() application(3 way conferencing)..?any idea about that...? * when asterisk make calls, if it is an answering machine or fax, what will be the response from asterisk server..? do we can drop the call accordingly...? * asterisk can sense the human voice...? * Do u have any idea how to create a soft phone like x-lite..? with regards, Robinson I guess you are very new to asterisk. Asterisk is very feature rich. Check out www.voip-info.org, it is a great resource about asterisk. Also find the book (on-line) 'Asterisk, the future of telephony' aka 'Asterisk TFOT', it will get you started on asterisk big time if you want to do the more intresting things as you describe. -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --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: OT (a little): IPV6 Ramifications Article
Hans Witvliet wrote: The only obstacles currently, are the ISP's. Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as an ipv4 address. afaik, all dsl-modems currently can only work with v4. (correct me if i'm wrong) So, let the modem be a modem, not a router, and do ip where it belongs, on the host/router. Now, for the more intresting questions, where to find a decent ipv6 firewall (yes bleeding edge linux kernels have one, sort of) Only option currently is: tunnelbroker.net Perhaps if we actually get directly fibre or ethernet to our home, ipv6 will get mainstream. hw (Well, if * would support it, without any extra patches, would help) And that is a problem, far to many new applications still get developed for ipv4 only, rather than using the generic interfaces available -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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] DISABLE 9?
JNA wrote: Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? the asterisk dialplan matches most specific entries first. So you could have one set for one or two ditgit internal numbers, one set for 7 digit local numbers, one set for 10 digit national numbers and one set for n digit international numbers all starting with an international prefix. ___ --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 -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] DISABLE 9?
Matt wrote: Have you never run into a situation where you dial +15705551212 for a number, but also have an extention of 157 or something? Of course, bu then again, a properly designed dialplan will have more specific entries for internal numbers _XXX which will match 157 but not 15705551212, so if you'd dial 15705551212, asterisk will have to find a less specific entry in your dialplan to match that to. The 9 is legacy, yes, but still important, in my opinion, to segregate the networks. You know that anything starting with a 9 is going to go outbound, and all of your extentions are then 1xx-8xx. 9anything is reserved for going to the PSTN. Otherwise, you are either going to have to have your callers dial 1areacode for everything (and then have your extentions 2xx-9xx), that is they can't just dial 5551212, which is a pain, or you are going to have overlap. The 9 may be legacy, but it is somewhat important! The 9 is legacy, american and IMNSHO completely obsolete. You can leave it in your dialplan as not to upset those users used to dialing a 9 for an outside line, but apart from that, it has no use anymore. -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --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
