Re: [Asterisk-Users] g729 license question
But I don't think Digium is in a hurry to implement such a feature since it forces people to buy more licenses than they really need to avoid dead calls. I don't think they're in ahurry either, but I doubt that whatever their commission on the $10/channel fee is has a big impact on their annual sales :) ___ --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] inform the agent about the queue he is answering
On 2/3/06, nik600 [EMAIL PROTECTED] wrote: On 2/3/06, Script Head [EMAIL PROTECTED] wrote: Yes, it is possible. You need to track the queue log and channels via manager console or by tailing logs in real time and then match the destination of the caller by the callerid. Then make the decision which URL to redirect the caller too. None of this comes with Asterisk but it is possible to build. hi i'm trying to tailing logs, this is the problem: 1139045971|1139045971.14|700|NONE|ENTERQUEUE||101 1139045978|1139045971.14|700|Local/[EMAIL PROTECTED],1|CONNECT|7 in the first row you can see that the extension 101 is entered in the queue 700 now, when the agents answer from extension Local/[EMAIL PROTECTED],1 the log reports the second row, but i need this information first than the answer of the agents! can i enable something in the queue logs due to see something like this? first log: 1139045971|1139045971.14|700|NONE|ENTERQUEUE||101 second log: ... . .. . . . . | 700 | ringing on 102 | 101 third log: 1139045978|1139045971.14|700|Local/[EMAIL PROTECTED],1|CONNECT|7 thanks ___ --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] g729 license question
On Sat, 2006-02-04 at 10:32 +0100, Wilson Pickett wrote: I don't think they're in ahurry either, but I doubt that whatever their commission on the $10/channel fee is has a big impact on their annual sales :) Their commission is about $9/channel according to pricing available at the registrar. I am looking at offering $5/channel licenses and other features, which includes site licenses (ie 1 channel for your site rather than locked to your mac addr) and some slack, becuase of one method I am looking at doing stuff it would be pooled, which would result in potentially less than $5/channel but also if you get 100 lcienses you could use upto 110 or something on occasion (ie not always, if you always need more you have to buy more). I am trying to offer some interesting stuff :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
This is probably a stupid question, but how do you specify multiple
fallovers? I.e., if provider1 is not reachable/busy, try provider2.
If provider2 is down, try provider3. If provider3 is down...etc. I
understand how to do it the old way, just keep adding 101 to the
extension. What would you add to a NOANSWER extension though? I
guess you could send it to a different context, then you could use
another NOANSWER, but I like keeping things short and easy.
Joseph Tanner
On 2/3/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is exactly the point what I am looking for. How can I use the next
peer in the dial logic? I was trying DIALSTATUS, ... but I could not
make it.
Should be easy; we use:
[macro-safedial]
;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4})
exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
exten = s-NOANSWER,2,Hangup
exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1})
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1})
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
Florian
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Re: [Asterisk-Users] callback script?
This is what I use, more or less: http://mundy.org/blog/index.php?p=73 , go down to Incoming Call Context (about 1/3 down). I had to modify it a bit, as I actually need Asterisk to pick up and listen to some DTMF digits before hanging up and calling me back, but it works great for me, and requires no external agi scripts. Joseph Tanner On 2/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote:
This is probably a stupid question, but how do you specify multiple
fallovers? I.e., if provider1 is not reachable/busy, try provider2.
If provider2 is down, try provider3. If provider3 is down...etc. I
understand how to do it the old way, just keep adding 101 to the
extension. What would you add to a NOANSWER extension though? I
guess you could send it to a different context, then you could use
another NOANSWER, but I like keeping things short and easy.
[snip]
[macro-safedial]
;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4})
exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
exten = s-NOANSWER,2,Hangup
exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1})
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1})
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
I have this macro too in my extensions.conf
Later in the dialplan I use:
[outgoing-speakup]
;dutch telephone nrs.
exten = _0X,1,Set(CDR(ACCOUNTCODE)=outgoing-speakup)
exten = _0X,2,Set(CALLERID(all)=X)
exten = _0X,3,Macro(safedial,${IAXTRUNK_SPEAKUP01}/31${EXTEN:1},50,Tr)
exten = _0X,4,Macro(safedial,${IAXTRUNK_SPEAKUP02}/31${EXTEN:1},50,Tr)
exten = _0X,5,Macro(safedial,${ZAPTRUNK}/${EXTEN},50,Tr)
exten = _0X,6,Congestion()
Works like a charm.
In my production environment I actually load balance calls
using RAND so both IAXTRUNK_SPEAKUP01 and IAXTRUNK_SPEAKUP02
get an equal load of calls, but that's not relevant to your
question :)
--
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http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the load balancing.
Regards,
Umair Bari
On 2/4/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote: This is probably a stupid question, but how do you specify multiple
fallovers?I.e., if provider1 is not reachable/busy, try provider2. If provider2 is down, try provider3.If provider3 is down...etc.I understand how to do it the old way, just keep adding 101 to the
extension.What would you add to a NOANSWER extension though?I guess you could send it to a different context, then you could use another NOANSWER, but I like keeping things short and easy.
[snip] [macro-safedial] ;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4}) exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3) exten = s-NOANSWER,2,Hangup exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1}) exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1}) exten = s-CONGESTION,1,Congestion exten = _s-.,1,Congestion exten = s-,1,CongestionI have this macro too in my
extensions.confLater in the dialplan I use:[outgoing-speakup];dutch telephone nrs.exten = _0X,1,Set(CDR(ACCOUNTCODE)=outgoing-speakup)exten = _0X,2,Set(CALLERID(all)=X)
exten = _0X,3,Macro(safedial,${IAXTRUNK_SPEAKUP01}/31${EXTEN:1},50,Tr)exten = _0X,4,Macro(safedial,${IAXTRUNK_SPEAKUP02}/31${EXTEN:1},50,Tr)exten = _0X,5,Macro(safedial,${ZAPTRUNK}/${EXTEN},50,Tr)
exten = _0X,6,Congestion()Works like a charm.In my production environment I actually load balance callsusing RAND so both IAXTRUNK_SPEAKUP01 and IAXTRUNK_SPEAKUP02get an equal load of calls, but that's not relevant to your
question :)--Michiel van Baakhttp://michiel.vanbaak.info[EMAIL PROTECTED]GnuPG key:
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2DWhy is it drug addicts and computer afficionados are both called users?___--Bandwidth and Colocation provided by
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Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
On 18:02, Sat 04 Feb 06, Umair Bari wrote:
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the
load balancing.
Regards,
Umair Bari
Umair,
Here is a actual copy/pasted block from my
[outgoing-speakup]
I have a block like this for dutch numbers, international
numbers, mobile numbers etc so they get another logentry
that I use in my cdr webtool.
;dutch telephone nrs.
exten = _0X,1,Verbose(1,Routing call from ${CALLERID(num)} to ${EXTEN}
on channel ${CHANNEL})
exten = _0X,2,Set(CDR(ACCOUNTCODE)=outgoing-speakup)
exten = _0X,3,Set(CALLERID(all)=XX)
exten = _0X,4,GotoIf($[${RAND(0,99)} + 50 = 100]?9)
exten = _0X,5,Macro(safedial,${IAXTRUNK_SPEAKUP01}/31${EXTEN:1},50,Tr)
exten = _0X,6,Macro(safedial,${IAXTRUNK_SPEAKUP02}/31${EXTEN:1},50,Tr)
exten = _0X,7,Macro(safedial,${ZAPTRUNK}/${EXTEN},50,Tr)
exten = _0X,8,Goto(12)
exten = _0X,9,Macro(safedial,${IAXTRUNK_SPEAKUP02}/31${EXTEN:1},50,Tr)
exten = _0X,10,Macro(safedial,${IAXTRUNK_SPEAKUP01}/31${EXTEN:1},50,Tr)
exten = _0X,11,Macro(safedial,${ZAPTRUNK}/${EXTEN},50,Tr)
exten = _0X,12,Congestion()
When I look at the console I see it indeed picks
IAXTRUNK_SPEAKUP01 and IAXTRUNK_SPEAKUP02 at random.
This is for 1.2.2 and later, that's when they moved the
Random() call to a dialplan function RAND
The only drawback I have is that if the other end is not
picking up I have to wait 150 seconds before I get a
Congestion ;) but I can live with that.
For that reason only the dutch numbers have the 3rd failover
to ZAPTRUNK
Hope this helps you a bit with understanding the RAND stuff.
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] 64bit processor and 32 bit digium card
On 2/4/06, Eduard B. Cleofe [EMAIL PROTECTED] wrote: Hi Guys, Im planning to setup a server that has a 64bit processor and 32bit digium card using 64bit kernel of Linux.Id like to know if il be having a problem later on its compatibility and the availability of drivers or patches for 64bit zaptel.Because we all know the 64bit are not much matured enough. Any comment and suggestion guys.Will highly appreciated. Thank you very much. We have clients with TE411P cards on 64 bit kernels and machines and have had no noticeable problems. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] No audio for outgoing calls
Hi, I've just noticed my Asterisk setup is having a small issue. - Whenever I get a call (from VoIP provider to my Asterisk box, forwarded to my GXP-2000 phone through SIP registration) I get perfectly clear audio, both ways. - When I call out with the phone (Phone to asterisk box through SIP registration, then to VoIP provider, than to PSTN to my home phone) I get NO audio. I know the one-way audio can be the fault of firewalls, but this is different (s I understand it). What could be causing this breakdown? Especially since when I get my phone last week, I made sure to call my home phone and I could hear perfectly. SO why can cause this sudden problem? Thanks, Mike ___ --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] Routing Calls via chan_capi with AVM FritzCard
Hello asterisk-users, I recently set up an asterisk server using Debian Sarge. I also added an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed a new kernel (2.6.15.1) with modular support for the CAPI stuff and also integrated the FritzCard driver available from AVM. capiinfo succeeds in correctly showing the ISDN card. When talking from one PC to another via SIP, everything works fine. But when trying to call the * server via ISDN, the call kind of doesn't get through: In the receiver, I hear nothing (no ringing tones at all), and a few seconds later, I get the line busy signal. Running capi debug on the asterisk console shows lots of debug output that I don't want to post here, but I always notice the following lines: ERROR[5832]: chan_capi.c:1197 find_pipe: unable to find a pipe for PLCI = 0x101 When we try it the other way around (routing an outgoing call from SIP via ISDN), the destination phone rings, but when picking up the line, it remains silent. Seems as if no voice signals get through. Unfortunately, Google couldn't help me. Maybe you can? My specifications: - asterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 - Debian Sarge on Linux 2.6.15.1 - FritzCard module built from the latest sources available by AVM My capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=3413 incomingmsn=3413 outgoingmsn=3413 controller=1 softdtmf=1 accountcode= context=isdn_incoming If you need any more information on my * configuration, please feel free to ask. Thanks in advance!! Regards, Christian Schmidt ___ --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] can asterisk to say chinese like say english
On Fri, Feb 03, 2006 at 11:32:32PM -0500, Wai Wu wrote: A better solution is write special modules for different language to say 1) a string of digits 2) numbers 3) currencies Translated into Asterisk jargon: patches adding support for Chineese into say.c would be welcomed. Luckily, HEAD seems to contain some support for the language zh in SayUnixTime and SayNumber . 1.2 doesn't, though. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Routing Calls via chan_capi with AVM FritzCard
Hi Christian, difficult to say for me. I would just recommend another config which runs stable on my i586-based embedded system: mISDN (latest CVS) and chan_mISDN (latest CVS as well) . I used this with a FRITZCard PCI and now switched to a HFC-S card and have tested that sucessfully in TE and NT mode ... Viel Erfolg Grüße, Jürgen Hello asterisk-users, I recently set up an asterisk server using Debian Sarge. I also added an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed a new kernel (2.6.15.1) with modular support for the CAPI stuff and also integrated the FritzCard driver available from AVM. capiinfo succeeds in correctly showing the ISDN card. When talking from one PC to another via SIP, everything works fine. But when trying to call the * server via ISDN, the call kind of doesn't get through: In the receiver, I hear nothing (no ringing tones at all), and a few seconds later, I get the line busy signal. Running capi debug on the asterisk console shows lots of debug output that I don't want to post here, but I always notice the following lines: ERROR[5832]: chan_capi.c:1197 find_pipe: unable to find a pipe for PLCI = 0x101 When we try it the other way around (routing an outgoing call from SIP via ISDN), the destination phone rings, but when picking up the line, it remains silent. Seems as if no voice signals get through. Unfortunately, Google couldn't help me. Maybe you can? My specifications: - asterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 - Debian Sarge on Linux 2.6.15.1 - FritzCard module built from the latest sources available by AVM My capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=3413 incomingmsn=3413 outgoingmsn=3413 controller=1 softdtmf=1 accountcode= context=isdn_incoming If you need any more information on my * configuration, please feel free to ask. Thanks in advance!! Regards, Christian Schmidt ___ --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] Routing Calls via chan_capi with AVM FritzCard
Christian Schmidt schrieb: [..] - asterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 Do your self a favour and get chan-capi_cm of Sourceforge http://sourceforge.net/projects/chan-capi -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Routing Calls via chan_capi with AVM FritzCard
On Sat, 4 Feb 2006, Christian Schmidt wrote: Hello asterisk-users, I recently set up an asterisk server using Debian Sarge. I also added an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed a new kernel (2.6.15.1) with modular support for the CAPI stuff and also integrated the FritzCard driver available from AVM. capiinfo succeeds in correctly showing the ISDN card. When talking from one PC to another via SIP, everything works fine. But when trying to call the * server via ISDN, the call kind of doesn't get through: In the receiver, I hear nothing (no ringing tones at all), and a few seconds later, I get the line busy signal. Running capi debug on the asterisk console shows lots of debug output that I don't want to post here, but I always notice the following lines: ERROR[5832]: chan_capi.c:1197 find_pipe: unable to find a pipe for PLCI = 0x101 When we try it the other way around (routing an outgoing call from SIP via ISDN), the destination phone rings, but when picking up the line, it remains silent. Seems as if no voice signals get through. Unfortunately, Google couldn't help me. Maybe you can? My specifications: - asterisk 1.0.7.dfsg.1-2 - chan_capi 0.3.5-11 You really should update to new chan_capi-cm version (you can find it on sourceforge.net). Armin - Debian Sarge on Linux 2.6.15.1 - FritzCard module built from the latest sources available by AVM My capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=3413 incomingmsn=3413 outgoingmsn=3413 controller=1 softdtmf=1 accountcode= context=isdn_incoming If you need any more information on my * configuration, please feel free to ask. Thanks in advance!! Regards, Christian Schmidt ___ --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] Fast AGI performance question
Good question, i would like to know the same. Im using MAGI patch to execute AGI commands via the Manager. I have a PHP proxy connected to the CallManager PHP server that do the routing stuff and decide to execute Dial, Voicemail, Playtones, receive DTMF or some other stuff in the channel, i have still not made hard tests, but it seems to be doing it fine for a couple of calls. I would like to know other people experience in similar circumstances. Eric: I do not know perl at all, how have you written the server, is multithreaded? should it be? Since PHP does not have threads my server is not, is a Event Driven server based on the manager events provided by MAGI patch. RegardsOn 2/3/06, Eric Lyons [EMAIL PROTECTED] wrote: I'm building a fast AGI application (server written in Perl using Net::Server), and have a sort of design performance question.Is fast AGI keeping the equivalent of a Manager API session open until it returns?Are there still deadlocking issues there (in1.2)?My assumption is that the fast AGI application -- which handles all incoming calls (_X. in dialplan context) -- should do itsbusiness (db lookups, setting channel variables, etc) as quickly as possible, then allow control to go back to asterisk in the dialplan for max performance.Certain applications would require staying in the call path for the duration of the call, and presumablythese would have more difficulty scaling?Eric.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --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 laptop connected to POTS line
Not a chance, they sell SPA3000's by the truckload. If you only need one line, then go with the SPA3000, if you need more, I would go with the Mediatrix 1204. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Saturday, February 04, 2006 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line I thought they stopped selling the spa3000 ? --- Damon Estep [EMAIL PROTECTED] wrote: Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 on laptop connected to POTS line
http://www.sipura.com/products/spa3000.htm -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Saturday, February 04, 2006 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line I thought they stopped selling the spa3000 ? --- Damon Estep [EMAIL PROTECTED] wrote: Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] ArtDio gateways
Does anyone have any experience (good or bad) with ArtDio gateways? I am having two problems, the configuration does not seem to be sticking (part does, part does not) and it ignores * commands from the phone. I checked and the phone is definitely sending the *. Thanks for you help [EMAIL PROTECTED] R C Schroeder ___ --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 can I configure to call from the consolebymeans of a sip phone,
It's something like exten = 15,1,Dial(Console/DSP) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Azzopardi Sent: Saturday, February 04, 2006 2:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How can I configure to call from the consolebymeans of a sip phone, I can call from the console by means of the 'dial' command, now I need to know how to call the console itself. Anthony. ___ --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] Routing Calls via chan_capi with AVM FritzCard
Hello Armin, Armin Schindler, 04.02.2006 (d.m.y): You really should update to new chan_capi-cm version (you can find it on sourceforge.net). OK, I gave that a try. Now, my server is running asterisk 1.0.10 with chan_capi-cm from SourceForge. When calling asterisk from my phone, it rings and rings and rings. Asterisk says: *CLI == 3413: Incoming call '0012341234' - '' Urgent handler == 3413: CAPI Hangingup Urgent handler The CAPI Hangingup occurs between the second and the third ring. It seems to me as if asterisk doesn't receive a destination msn. Unfortunately, I can only access the asterisk server but not the PBX that provides my ISDN channels... At the moment, I cannot test if voice gets through when doing outgoing calls via CAP (I'm at home, and the server is located at my department). Thank you very much!! Gruss, Christian Schmidt -- Letzte Worte eines Bombenlegers: Was iss'n das für'n Draht? ___ --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] a couple of questions
Well, I posted this question a few days back and got no answers, I
just figured it out, so here's the answer to part of it and maybe
someone can still answer the how to I get the SIP extension that
called this macro part.
exten = _6[0-2][0-4],1,Flash()
exten = _6[0-2][0-4],2,Dial(SIP/1${EXTEN:1},,rtT)
Is there any way to get the SIP extension that called this macro?
I've also tried:
in extensions.conf
[context]
exten = s,12, set(DYNAMIC_FEATURES=zapflash)
exten = s,13,dial(${zaino_in},400,tTj)
in features.conf
[applicationmap]
zapflash = *3, callee, flash ; does not work
zapflash=*3,callee,flash; Like this it works perfect
* is very sensitive to spaces, as soon as I removed all of the spaces
in the zapflash line everything started working correctly. I've
always pus spaces after commas for readability, but it doesn't always
or ever work with *.
Ira
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[Asterisk-Users] Visio-type symbol for an Asterisk/VoIP server?
I was wondering if anyone knows whether or not there is an accepted icon for a telephony server for use in diagramming programs like Visio/Dia/etc. The Cisco set has an icon for an IP phone, but I can't find one for a telephony server. I'm sure there must be such for telephone switches too, but maybe computer-based servers ought to have their own. It would be cool to have a generic one for a telephony server, and then a custom version (like with a * symbol in or around it) for Asterisk. Unfortunately, all my fingers are good for are typing characters. Anyone know of such a thing, or want to sketch one up? Thx. B. ___ --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 laptop connected to POTS line
Do people not use the Grandstream ATA's because they are cheap or because there is actually a problem with them? They have a 2 line version for around $50 that I have used in various locations. I have about 8 or so. They seem to do an excellent job. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Saturday, February 04, 2006 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line Not a chance, they sell SPA3000's by the truckload. If you only need one line, then go with the SPA3000, if you need more, I would go with the Mediatrix 1204. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Saturday, February 04, 2006 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line I thought they stopped selling the spa3000 ? --- Damon Estep [EMAIL PROTECTED] wrote: Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Routing Calls via chan_capi with AVM FritzCard
On Sat, 4 Feb 2006, Christian Schmidt wrote: Hello Armin, Armin Schindler, 04.02.2006 (d.m.y): You really should update to new chan_capi-cm version (you can find it on sourceforge.net). OK, I gave that a try. Now, my server is running asterisk 1.0.10 with chan_capi-cm from SourceForge. When calling asterisk from my phone, it rings and rings and rings. Asterisk says: *CLI == 3413: Incoming call '0012341234' - '' Urgent handler == 3413: CAPI Hangingup Urgent handler The CAPI Hangingup occurs between the second and the third ring. It seems to me as if asterisk doesn't receive a destination msn. What kind of connection type is that? MSN without a number is unknown to me. Anyway, you would need to set your extensions.conf to accept 'no number'. And you might need to set immediate=yes in capi.conf Armin ___ --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: 5,000 concurrent calls system rollout question
On Thu, 2 Feb 2006, John Todd wrote: [SNIP] 3) Nobody else has thus far taken the bait and made any comments about their systems. I appreciate Signate's comments; they seem to be the only ones to publicly claim large-scale throughput using Asterisk in a public forum. Most other people who claim thousands or even high hundreds of connections do so offhand, without responding to second questions when I raise my figurative eyebrows. John, Per our conversation in San Fransciso, I am starting to push a couple of my Asterisk boxes farther than I've gone before. I'm not yet anywhere near the 5,000 concurrent call level on my boxes, but I am starting to see 150-160 concurrent calls coming through the system. In this case, these are SIP to SIP where Asterisk is staying in the media stream, but rarely transcoding. Approximately 99% of the calls coming through are just pass-through g729, with the occasional gsm conversion. I'm running Asterisk 1.2.4-svn in a completely stock configuration. I.E. no patches whatsoever, and absolutely performance tweaks. In fact, the system is running using MALLOC_DEBUG to catch memory leaks and is built using dont-optimize so we can get backtraces if things go south. My Dial-Plan is highly optimized, with a focus on being as efficient as possible while offering failover options for call completion. 4) There are still no notes on other problems with scale here. I've had systems with several hundred simultaneous SIP connections, but sip show channels sure does start to take a while. What _other_ problems crop up, but don't necessarily cause a failure condition? Well, debugging anything on the console with 160 concurrent calls coming through the system (sometimes 4-5 calls / second) is nearly impossible. Most of the time, I don't even run the console, and simply execute commands from a bash prompt as asterisk -rx 'sip show channels'. I ALWAYS, ALWAYS, ALWAYS issue a set verbose 0 before I reload the box, as a reload causes the box to hiccup slightly while it is printing the data to the console. I had originally opted to write CDRs to disk and then import them into a SQL database, but after I cleaned up my dial-plan, I opted to use cdr_odbc. I am concerned that this could cause a blocking condition if the SQL server is unavailable, but for now I'm taking the risk because I need to have real-time stats on call statistics. 5) I will agree that most SIP testing systems are currently too pricey. I would love to find a well-connected network that rents out a few of the better-known SIP testing tools to beat on Asterisk installations in remote places for short periods of time. But this has always been the case... test gear is a small market, and expensive. Just look at the MSRP of new high-end HP Oscilloscopes if you want to get a picture of price-gouging. I know that Olle spent some time at SipIt w/ Asterisk, and he's been interested in doing some additional compliance and scalability testing. I'd like nothing better than to get a couple of key developers together for a weekend of scalability bashing somewhere, preferably outside of the regular conference circuit (too distracting) to push things to their limits. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ --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] Regarding cdr_manager.conf
Victor Alvarez wrote: Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? the cdr_manager.conf file control weather the Asterisk manager should include the cdr event. it has nothing to do with Master.csv file. (you can read about Asterisk manager here: http://www.voip-info.org/wiki/view/Asterisk+manager+API ) to enable the cdr engine in general, set enable = yes in cdr.conf and setup at least one of .conf files specific to different databases: Master.csv - cdr_custom.conf Mysql - cdr_mysql.conf odbc - cdr_odbc.conf postgreSQL - cdr_pgsql.conf FreeTDS - cdr_tds.conf -- __ Edwin Lam [EMAIL PROTECTED] __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370 __ __ http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xEF11A895 __ ___ --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] g729 license question
Please let me know when you are going to do it. My clients typical requirement is a few hundred license. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter aka Bret McDanel Sent: Saturday, February 04, 2006 4:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] g729 license question On Sat, 2006-02-04 at 10:32 +0100, Wilson Pickett wrote: I don't think they're in ahurry either, but I doubt that whatever their commission on the $10/channel fee is has a big impact on their annual sales :) Their commission is about $9/channel according to pricing available at the registrar. I am looking at offering $5/channel licenses and other features, which includes site licenses (ie 1 channel for your site rather than locked to your mac addr) and some slack, becuase of one method I am looking at doing stuff it would be pooled, which would result in potentially less than $5/channel but also if you get 100 lcienses you could use upto 110 or something on occasion (ie not always, if you always need more you have to buy more). I am trying to offer some interesting stuff :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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] Maximum retries exceeded on call/phantom calls?
I am confused due a side effect produced in my * installation. It consists of 1 Sangoma A101 E1/lSDN PRI card connected to Telmex service 16 analog phones thru SIP enabled SP5004 Micronet gateways 4 SIP hard phones. Everything in a local network/no natting. We are processing nearly 2000 calls/day outgoing/incoming Everithing seems to be ok but after an hour or so I begin to see the message Maximum retries exceeded on call... On my logging console. This message continues to appear with a climbing frequency on different call ids till the entire system begin to unregister my sip clients. Asterisk needs to be restarted as if it has suffered a DOS attack. Prior to this situation arrives, I notice that phantom calls rings phones but nobody there--- After a couple of weeks of debugging I notice that this situation could be related to 3-way calling from the operator to other sip extensions. This tranferred calls seems not to die after the normal operation of the feature (flash/get tone/dial extension/speak with employee/hangup). I have all my sip gateways set to support transfer, so SIP attended transfer is done by the gateway and by zapata at the same time producing the side effect? Waiting some feedback OCA -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.2/251 - Release Date: 04/02/2006 ___ --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: delaying answer for a number of rings or an amount
It sounds like you both need a Zap card. You can ring the analog phone and/or the Sip phones when a call comes in on the POTS line that is connected to the card. MARK. Brian J. Murrell wrote: On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote: Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. Ahhh. So we share the latter at least. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Hrm. Yeah, this is what I'm trying to do. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. So you don't want Asterisk to wait and see if the POTS line is picked up before ringing the SIP phones? Interesting. 2 - After 30 seconds if the line is still ringing (nobody picked up POTS phone or SIP phones) * answers the line and sends to Voicemail. Asterisk never picks up the call until the 30 seconds are up. What seems to be happening here is that even if somebody picks up the POTS line within a few seconds, after the 30 seconds (Wait() in my case, but I'd imagine the same will happen after ringing the SIP lines for 30s) is up Asterisk is also on the POTS line (with the callee who picked up the POTS phone) doing the voicemail intro and recording the conversation. [from-pots] exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) exten = s,3,Hangup I will try this exactly and see if it works any better. b. ___ --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] Routing Calls via chan_capi with AVM FritzCard
Hello Armin, Armin Schindler, 04.02.2006 (d.m.y): On Sat, 4 Feb 2006, Christian Schmidt wrote: OK, I gave that a try. Now, my server is running asterisk 1.0.10 with chan_capi-cm from SourceForge. When calling asterisk from my phone, it rings and rings and rings. Asterisk says: *CLI == 3413: Incoming call '0012341234' - '' Urgent handler == 3413: CAPI Hangingup Urgent handler The CAPI Hangingup occurs between the second and the third ring. It seems to me as if asterisk doesn't receive a destination msn. What kind of connection type is that? MSN without a number is unknown to me. Well, I'm not that familiar with telephony stuff, but our ISDN line comes from a bigger PBX (university department). Anyway, you would need to set your extensions.conf to accept 'no number'. Could you give me a hint on how to do that? I already tried defining rules for the extension ., but that did't work. And you might need to set immediate=yes in capi.conf OK, thank you! Regards, Christian Schmidt -- Was die neuen Unwissenden holen müssen: Gewichte für die Wasserwaage ___ --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] ddi???
You need to get BT to agree and allocate or port the numbers. You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) I don't know the number of digits that BT pass through on a PRI, but on a set of BRIs with a range of DDIs, they're passing the last 6 digits (so given the OP's range, you'd want to match on 838381 etc.) I concur with Tim's suggestion of trying to get the internal extensions related to the DDIs - it'll simplify your dialplan substantially. Out of curiosity, why do you want to go to BT for the number range? 8 channels through BT will cost a small fortune, and you could run 8 concurrent calls over a standard ADSL connection in the UK with appropriate codec selections. There are at least 3 or 4 companies in the UK that'll offer you a consecutive number range for a UK area code. You'd also avoid a substantial chunk of potential echo issues. The asterisk deployments we've done where the client has had calls delivered via IAX from a provider have all been *much* easier and taken far less time than when we have to fight with ISDN lines, or worse, analogue lines. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] hardware and network requirements
i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000 One of our clients has a similar sized setup running on an Athlon64 2800+ (2.2Ghz I think), 1GB RAM, 2x80GB HDDs in RAID1. You don't say how the calls are coming in, but I'd try and keep transcoding to a minimum. if they're coming from a PRI (i.e. alaw or ulaw) and you want to keep them that way down to the users, 25 concurrent calls @ 80kbps-ish is only 2mbps, so even a 100mbps LAN is fine for the task. Personally, I build our asterisk boxes rather than buying off-the-shelf servers, but I doubt it makes much difference one way or t'other. Go with whichever approach you feel most comfortable. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --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] ddi???
On Sat, 2006-02-04 at 23:33 +, Chris Bagnall wrote: You need to get BT to agree and allocate or port the numbers. You need to agree how many digits BT will pass on to you (probably 1925838395 but possibly just the last 2) I don't know the number of digits that BT pass through on a PRI, but on a set of BRIs with a range of DDIs, they're passing the last 6 digits (so given the OP's range, you'd want to match on 838381 etc.) BT generally like to pass 6 digits in and out of their network. You can request to have fewer digits sent; I'm not sure if they will let you have more. p. ___ --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] No audio? Update your Asterisk
Roger Hill wrote: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? If you are under linux rm /usr/lib/asterisk/modules/* rm /usr/include/asterisk/* cd asterisk-1.2.4 make clean make upgrade asterisk -r stop now safe_asterisk that's all Sergio ___ --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] DTMF Sporadicaly Being Generated
I've been having horrible DTMF problems lately on from Sipura ATAs to ZAP and IAX. It's primarily with repeated digits. I'm starting to move my connections to SIP until I can get it all figured out. Other than updating to the newest SVN trunk I haven't made changes on my end that should have caused this. I've already put some of my IAX debug on a bug report relating to double dtmf with Jitterbuffer enabled. MARK. Kevin P. Fleming wrote: Michael L. Young wrote: I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I was using before upgrading to the TE411P. This is a known problem, been discussed on the lists many times. You should contact Digium Support, since you just purchased a Digium card. They are best equipped to handle issues related to Digium hardware. ___ --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] DTMF Sporadicaly Being Generated
To quote Kevin: DTMF handling in the trunk is in a state of flux right now. It won't be resolved until this weekend. Don't use SVN for a production system, it's lots broken right now. If you really must, stick with r8786 for a while. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Hulber Sent: Sunday, 5 February 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF Sporadicaly Being Generated I've been having horrible DTMF problems lately on from Sipura ATAs to ZAP and IAX. It's primarily with repeated digits. I'm starting to move my connections to SIP until I can get it all figured out. Other than updating to the newest SVN trunk I haven't made changes on my end that should have caused this. I've already put some of my IAX debug on a bug report relating to double dtmf with Jitterbuffer enabled. MARK. Kevin P. Fleming wrote: Michael L. Young wrote: I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I was using before upgrading to the TE411P. This is a known problem, been discussed on the lists many times. You should contact Digium Support, since you just purchased a Digium card. They are best equipped to handle issues related to Digium hardware. ___ --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] DTMF Sporadicaly Being Generated
Good to know. I was able to play around and get it mostly working but I'm still not able to get DTMF working with Jitterbuffer ON for IAX although I previously could at least with some providers. I had to define my SIP extensions to use INBAND and set the Sipura devices to also use INBAND and not process INFO or AVT. I also noticed I was using dtmf= instead of dtmfmode= which may or may not be in my imagination that the latter works better (or at all). Now at least people can listen to voicemail and authenticate a remote conference call using the same device. MARK. Rob Thomas wrote: To quote Kevin: DTMF handling in the trunk is in a state of flux right now. It won't be resolved until this weekend. Don't use SVN for a production system, it's lots broken right now. If you really must, stick with r8786 for a while. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Hulber Sent: Sunday, 5 February 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF Sporadicaly Being Generated I've been having horrible DTMF problems lately on from Sipura ATAs to ZAP and IAX. It's primarily with repeated digits. I'm starting to move my connections to SIP until I can get it all figured out. Other than updating to the newest SVN trunk I haven't made changes on my end that should have caused this. I've already put some of my IAX debug on a bug report relating to double dtmf with Jitterbuffer enabled. MARK. Kevin P. Fleming wrote: Michael L. Young wrote: I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I was using before upgrading to the TE411P. This is a known problem, been discussed on the lists many times. You should contact Digium Support, since you just purchased a Digium card. They are best equipped to handle issues related to Digium hardware. ___ --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 ___ --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] click to talk
--- Graziano Poretti [EMAIL PROTECTED] wrote: any idea where i can find the sip client to embed in my website ? (c# - java or whatever) SIP: http://www.vaxvoip.com/WebDemo/Softphone.HTM http://www.microappliances.com/site/html/index.php http://www.etntalk.com/callto/loginany/ http://www.worksoutsoft.com/products/intellIPhoneSDK.aspx IAX: http://www.silicontechnix.com/webtelefone/start.html __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Difference between VoiceMail and VoiceMail2?
Can someone explain the difference between VoiceMail and VoiceMail2? Thanks! ___ --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] G729a Pass-Through and Recording/Monitoring
Digium confirmend that this was still the case but trixter may have a way to at least make things much more efficient and save alot of money, especially in a recording situation. See his announcemnt here. http://www.trxtel.com/index.php?page=G_729_Codec Thanks, Steve Totaro -Original Message- From: Adam Goryachev Sent: Tue 1/24/2006 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] G729a Pass-Through and Recording/Monitoring On Mon, 2006-01-23 at 12:16 -0500, Steve Totaro wrote: Is this also true for recording of calls? Will I require licensing for each recorded call? Will the server see a big performance hit in this setup whether or not a license is required? In my experience (which was using asterisk 1.0.x at the time) you will need: 1 license to decode the audio and send to your output (PSTN as alaw) 1 license to decode the audio and write to disk as gsm Which seemed rather . non-optimal... perhaps that has changed in the past 6 months or more though :) Regards, Adam ___ --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 winmail.dat___ --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] Monitoring
You could use big brother or something. -Original Message- From: [EMAIL PROTECTED] Sent: Fri 1/27/2006 7:04 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitoring Hi asterisk and ser users, Is there a solution to monitor asterisk and ser with snmp ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 winmail.dat___ --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] G729a Pass-Through and Recording/Monitoring
On Sat, 2006-02-04 at 22:01 -0500, Steve Totaro wrote: Digium confirmend that this was still the case but trixter may have a way to at least make things much more efficient and save alot of money, especially in a recording situation. See his announcemnt here. http://www.trxtel.com/index.php?page=G_729_Codec Thanks :) Only encoding and/or decoding requires a license. If you are just pushing bits you dont need a license. This is according to http://www.sipro.com the people who do the licensing for G.729. There is no difference in a license for decode only or encode only vs both. They also do G.723.1 licensing and with G.723.1 there is a difference in licensing cost for decode only or encode only vs both. So you would see a savings if you were writing an app that only recorded for example. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] G729a Pass-Through and Recording/Monitoring
The original quesiton was that if you had a server performing G729 passthrough, could you do recording without licensing. Digium confirmed that the server doing the passthrough would also need a license in order to record the conversation. Using the Erlag formula you can pretty much figure out how many licenses would be needed. My situation is two locations operating as a single logical call center. G729 is good for the bandwidth and works fine with the Tenor boxes I plan to use. Obviously, not all 672 channels would be in use at a time but it could be possible and obviously not all phone calls will need to be recorded. Planned properly, this could amount to significant savings. Thanks, Steve -Original Message- From: trixter aka Bret McDanel Sent: Sat 2/4/2006 10:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] G729a Pass-Through and Recording/Monitoring On Sat, 2006-02-04 at 22:01 -0500, Steve Totaro wrote: Digium confirmend that this was still the case but trixter may have a way to at least make things much more efficient and save alot of money, especially in a recording situation. See his announcemnt here. http://www.trxtel.com/index.php?page=G_729_Codec Thanks :) Only encoding and/or decoding requires a license. If you are just pushing bits you dont need a license. This is according to http://www.sipro.com the people who do the licensing for G.729. There is no difference in a license for decode only or encode only vs both. They also do G.723.1 licensing and with G.723.1 there is a difference in licensing cost for decode only or encode only vs both. So you would see a savings if you were writing an app that only recorded for example. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group winmail.dat___ --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] G729a Pass-Through and Recording/Monitoring
On Sat, 2006-02-04 at 22:44 -0500, Steve Totaro wrote: The original quesiton was that if you had a server performing G729 passthrough, could you do recording without licensing. Digium confirmed that the server doing the passthrough would also need a license in order to record the conversation. Using the Erlag formula you can pretty much figure out how many licenses would be needed. The real answer is maybe. If you record raw g.729 you dont need a license becuase you arent encoding or decoding. However monitor may not work this way, it may internally decode even if it doesnt have to, I havent looked so I dont know. You would then only need a license to change the coding scheme (ie from g.729 to anything else) or play the file (whatever plays it at the very least would need a license). In this model you could record raw g.729 frames, and have 1 process, thus 1 license to convert them to something else. But the issue of whether or not monitor would decode (or by using monitor cause something else to decode) would need to be resolved. ranchnetworks.com has network appliances that work with asterisk. These have a calea feature, which basically does port replication on the individual RTP streams that are flagged (ie not everything). It works in two modes, one it sends a copy of the RTP data to a specified IP/port or it just replicates and you can use a packet sniffer. Either mode would enable you to cleanly record without a license, see above for listening. This also assumes that there is traffic going through their switch, becuase well if it doesnt its a little hard for their switch to do anything with it :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] hardware and network requirements
Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz, 1GB RAM,I'm doing no transcoding btw.Alyed Return-Path: [EMAIL PROTECTED] Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 4 Feb 2006 16:59:29 -0700 i'm planning to migrate a callcenter to asterisk and VOIP, the call center can have up to 25 cuncurrents agents logged in. Can a normal server with 1 GB ram 100 GB HDD Pentium 4 3.6 Ghz CPU Ethernet 10/100/1000One of our clients has a similar sized setup running on an Athlon64 2800+(2.2Ghz I think), 1GB RAM, 2x80GB HDDs in RAID1.You don't say how the calls are coming in, but I'd try and keep transcodingto a minimum. if they're coming from a PRI (i.e. alaw or ulaw) and you wantto keep them that way down to the users, 25 concurrent calls @ 80kbps-ish isonly 2mbps, so even a 100mbps LAN is fine for the task.Personally, I build our asterisk boxes rather than buying off-the-shelfservers, but I doubt it makes much difference one way or t'other. Go withwhichever approach you feel most comfortable.Regards,Chris-- C.M. Bagnall, Director, Minotaur I.T. LimitedThis email is made from 100% recycled electrons___--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] early media
Hi,all Does asterisk support sip early media? I have a setup asterisk for sip ATA boxs and a SIP trunk (SIP GATEWAY) for PSTN access. The ATA can call PSTN phone, cell phone, BUT it cant receive early media. I am sure the SIP GATEWAY support early media. If use the ATA connect to the gateway directly, it can receive early media. Jiangzhou Best Regards ___ --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] Search for Links for Communicating PC to PC in the same lan through Asterisk
Hi I am trying to do some simple experiment with Asterisk . my intention is to communicated two PC in my lan to voice -communicate with each other with out extra hardware I searched the FAQ and wiki for any links for this , so far I have not found one , It would be much help , if I get a link on communicating PC to PC in the same lan through Asterisk Thanks Joseph John ___ Yahoo! Photos NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com ___ --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
