Re: [asterisk-users] {top|bottom|interleaved} posting, once again
Steve Edwards schrieb: On Sat, 6 Feb 2010, Michelle Dupuis wrote: [snip] Oh wait, the advent of computers has allowed us to conveniently insert the most recent text at the TOP of a message, to prevent people from having to reread the same stuff every time. A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? Just because you can do something with a computer doesn't mean you should. Just because this list's moderator has chosen bottom posting doesn't make it right, logical, common sense, etc. (These lists are not moderated...) The list owner's choice did not make bottom posting right, logical and common sense. It was all of those things already. The list owner just made the right choice. How about we don't belittle people who don't notice? (I don't think I belittled anyone for top-posting.) We all make mistakes, but once we are informed, continuing to make the same mistakes indicates either a lack of consideration or stupidity or both. Simple courtesy would be to only include relevant sections of previous posts and to reply below the quoted text. If you don't like the rules of the playground, find another playground. Actually bottom-posting without trimming anything (SCNR) is about as annoying as top-posting. Interleaved posting is fine, quoting just enough text so everyone can understand the context. But I have almost given up on this endless fight. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] large scale paging
Mark Willis schrieb: Has anyone done any large scale intercom deployments with Asterisk? I've been asked about building a system to one-way page 500 phones simultaneously from a single server. My concerns are: - My limited math capabilities suggest 41 Mbps of RTP traffic, which seems like a lot, plus asterisk would be taking a single input stream and exploding it out to 500 endpoints. - There are 500 near-simultaneous INVITEs being sent. Can the SIP channel handle that? Any suggestions or war stories are appreciated. Multicast RTP might be the solution. http://wiki.snom.com/Settings/multicast_listen http://wiki.snom.com/Settings/multicast_address http://forum.snom.com/index.php?showtopic=1905 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Snom M3s
--[ UxBoD ]-- schrieb: does anybody know how to reboot a SNOM M3 base station remotely ? wget --user='admin' --password='admin' \ 'http://snom-m3-ip-address/reboot.html' Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
Kevin P. Fleming schrieb: Rick Green wrote: 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! Good idea, done! A big thank you! Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DEVICE_STATE
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,NoOp(0317998...@inputinterior.se has state
${DEVICE_STATE(SIP/0317998975)})
exten = 0317998975,2,Dial(SIP/0317998975)
It doesn't matter if I have a call on 0317998975 or not. i always get:
-- Executing [0317998...@inputinterior.se:1]
NoOp(SIP/0317998985-0011, 0317998...@inputinterior.se has state
NOT_INUSE) in new stack
So I figure out that I have missed something but cant figure out what. :(
Any ideeas?
sip.conf:
[general]
allowsubscribe = yes
notifyringing = yes
notifyhold = yes
limitonpeers = yes
Philipp Kempgen
--
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Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Zeeshan Zakaria schrieb: I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions extensions == sip.conf peers? are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. They do, if rtcachefriends is enabled in sip.conf. Is there any work around? On voip-info I read that Snom phones can use BLF without using hints. Huh? Is it possible to do similar on Aastra phones? Carlos Chavez schrieb: You need to enable rtcachefriends=yes in sip.conf Zeeshan Zakaria schrieb: It is already enabled in sip.conf. All I can say is that it should work. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterik act as a SIP Proxy
Gayathri G schrieb: 1. Can I use Asterisk as a SIP Proxy. ( I want it to act as proxy not a B2b/GW) No. Asterisk is a back-to-back user agent (B2BUA). You might want to have a look at http://en.wikipedia.org/wiki/OpenSER Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Email
Thomas Perron schrieb:
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application generally
echo body of message .?
Please read the man page of the `echo` command.
man echo
( http://unixhelp.ed.ac.uk/CGI/man-cgi?echo )
mail -s --what does this do please?
subject line .comes from where?
man mail
( http://unixhelp.ed.ac.uk/CGI/man-cgi?mail )
${the_caller_...@tmobile.net) i understand this part.
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
And, then send an email to the party. Example
3035551...@tmobile.net
Summary
1. Capture the CID number.
2. Prepend his number to his service provider SMTP address
3. Email it to his phone
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
Note the usage of '|' here. IIRC it needs to be escaped on Asterisk
1.4.x and below.
I assume I need to install SendMail and play around with CID stuff.
Sendmail, postfix, exim, qmail - any program that provides a local
sendmail interface.
I personally prefer postfix.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] How to use SIP hints and BLF for realtime extensions on Aastra phones?
Zeeshan Zakaria schrieb: Actually BLF works fine on one system with Asterisk 1.4.22-4 but not on the other one with Asterisk 1.4.18. Both have exact same sip.conf and extensions.conf, same extension numbers. Is there anything else which could effect it. The one on which it doesn't work is on a virtual machine, on a virtual network. Try to narrow down the problem. I.e. try Asterisk 1.4.22-4 on a virtual machine or Asterisk 1.4.18 on physical hardware. On Sun, Dec 6, 2009 at 11:07 AM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Zeeshan Zakaria schrieb: I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions extensions == sip.conf peers? are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. They do, if rtcachefriends is enabled in sip.conf. Is there any work around? On voip-info I read that Snom phones can use BLF without using hints. Huh? Is it possible to do similar on Aastra phones? Carlos Chavez schrieb: You need to enable rtcachefriends=yes in sip.conf Zeeshan Zakaria schrieb: It is already enabled in sip.conf. All I can say is that it should work. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get Queue values from dialplan (Was: queue_variables() function)
Olivier schrieb:
2009/12/4 Olivier oza-4...@myamail.com
Has someone successfully used this QUEUE_VARIABLES() function (in
1.6.2-rc7) ?
A previous question about it remainded unanswered (
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466).
http://lists.digium.com/pipermail/asterisk-users/2009-February/227122.html
http://lists.digium.com/pipermail/asterisk-users/2009-February/227127.html
https://issues.asterisk.org/view.php?id=14506
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] Get Queue values from dialplan (Was: queue_variables() function)
Olivier schrieb:
2009/12/4 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
How can can you get current queue's length (ie maxlen) or waiting call
number from dialplan ?
Set(err=${QUEUE_VARIABLES(techsupport)});
Verbose(1,maxlen: ${QUEUEMAX});
Verbose(1,waiting calls: ${QUEUECALLS});
When includiing in my dialplan the same lines as yours, QUEUEMAX value
remains empty (while err equals -1).
With CLI, queue show techsupport says something like :
techsupportl has 0 calls (max 3) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
SIP/109 (Not in use) has taken no calls yet
No Callers
I also tried with and without setinterfacevar=yes or setqueuevar=yes.
Did you try with 1.6.2 ?
Can't remember. Maybe I tested this with 1.6.0 or 1.6.1.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky schrieb: BTW, meantime I have alread implemented all that. My DNS server is up running. I've chosen one of the existing registrars and payed him for registering 7.6.5.4.3.2.1.1.3.4.e164.arpa as my number at nic.at. The registrar vaildated that this really is my number, and when this was confirmed, did the registration. Then I registerd my DNS server as the authorative master for the domain *.7.6.5.4.3.2.1.1.3.4.e164.arpa Where exactly did you register your DNS server? Did your registrar handle it for you? http://www.nic.at ? http://www.enum.at ? That's it. It works! That's good news! When ever anyone anywhere in the world does a ENUMLOOKUP(mynumber), my server receives a request and (hopefully) sends the correct answer. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Leif Neland schrieb: Norbert Zawodsky skrev: The number +43-1-3207978 is my telephone number. I own it as long as I pay for it. And with extra digits behind it I can do whatever I like. I can create any extension - physical or virtual. I can attach a phone to extension 12, attach a virtual fax server for extension 12 to extension 99912 or could fire up my toaster if I call extension 911. I can invent any numbering scheme for my company. That's a fact! Again - At least here in Austria !! (can't speak for other countries) Invent all you want, nobody can call those fantasy-numbers anyway. Perhaps, a fraction of a percent, who are using ENUM. Leif, ever heard of direct inward dialing and PRI? http://en.wikipedia.org/wiki/Direct_inward_dialing http://en.wikipedia.org/wiki/Primary_rate_interface You can actually own a block of numbers like 01234567. You are free to map these DID numbers to extensions or do what ever you like. And it is guaranteed that nothing in the 01234567... range will ever be assigned to a different PSTN subscriber. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parsing custom SIP headers
Leif Madsen schrieb: Philipp Kempgen wrote: Just to be sure: Is there a dialplan function in Asterisk that parses custom name-addr-style SIP headers for me? Try this: https://issues.asterisk.org/view.php?id=16268 Thanks but I don't see the connection. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parsing custom SIP headers
Hi,
Just to be sure: Is there a dialplan function in Asterisk that
parses custom name-addr-style SIP headers for me?
If I wanted to do it right the syntax
name-addr *(SEMI generic-param)
is quite complex to parse in the dialplan using nothing but CUT().
It's so easy to make false assumtions about angle brackets ( ),
whitespace (LWS), quotes () around the display-name, character
escaping etc. All of the applications of CUT() I have seen are
way too simplistic.
Example of how it could work:
Set(addr=${SIP_PARSE_HEADER(${SIP_HEADER(P-Asserted-Identity)},addr-spec)});
Interesting parts include:
name-addr, display-name, addr-spec, scheme, userinfo, user,
telephone-subscriber, host, hostname, port, ...
Actually headers like P-Asserted-Identity can even have more then
one value.
---cut---
PAssertedID = P-Asserted-Identity HCOLON PAssertedID-value
*(COMMA PAssertedID-value)
PAssertedID-value = name-addr / addr-spec
---cut---
so I guess SIP_PARSE_HEADER() would need an index argument, just
like SIP_HEADER().
Proper parsing could be done in an AGI() script of course but that
involves a big overhead especially since the code to parse name-addr
is already in Asterisk. It's just not available in the dialplan.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)
bilal ghayyad schrieb: To be able run Asterisk and gnugk on the same machine at same IP address, I need to know how to configure the port ranges of the (Q931, H245, T120, RTP) for the asterisk H323 channel to avoid any confilict with the gnugk? From where to determine these ranges? About gnugk, I know from where to determine it, but I do not know how to determine these port ranges in the Asterisk H323. Not really an answer to your question but why not simply use different IP addresses? (bindaddr) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI stuff
Thomas Perron schrieb: How do I get to this prompt? #!/usr/bin/php -q ?php http://en.wikipedia.org/wiki/Shebang_%28Unix%29 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone/debug panel with BLF
Leif Neland schrieb: Philipp Kempgen skrev: Leif Neland schrieb: Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? X-Lite? It does not subscribe to hints on Asterisk. It does. In the contact drawer: Add contact - Contact Methods: Softphone, Phone/Address = Extension, tick Show this contact's availability. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone/debug panel with BLF
Leif Neland schrieb: Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? X-Lite? http://www.counterpath.com/x-lite.html Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Database postgresql not able to start
Tzafrir Cohen schrieb: On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in wrote: THIS IS MY /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF Small hint: Text in ALL CAPS is generally considered as shouting. Please try to avoid that if you don't really need it. Doug Lytle schrieb: aster...@opensourcesolution.in wrote: *this is my /var/lib/pgsql/data/postgresql.conf* In my client, it didn't show in all upper case (SeaMonkey) It actually was all-uppercase in the text/plain part. Doug, you are using the built-in HTML-to-text converter. | Content-Type: text/plain | | THIS IS MY | /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF | Content-Type: text/html | | pnbsp;strongthis is my /var/lib/pgsql/data/postgresql.conf/strong/p aster...@opensourcesolution.in: Please get a better mail client. Tzafrir Cohen schrieb: On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in wrote: # CONNECTIONS AND AUTHENTICATION I guess it is OK to use all-caps here as this text was copied from the config file. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No dahdi_zttools in AsteriskNow?
Humanx2000 schrieb: Just picked up Asterisk the Future of Telephony, every other listed program is there (Book does not tell you about the changeover to dahdi_toolname). But there is no dahdi_zttools. I have dahdi-tools installed, tried to install via yum and says it is already installed. zttool is now called dahdi_tool. dahdi_tabtab Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Import
Khaled W Chehab schrieb: how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql cdr_mysql from Asterisk Addons. Configuration file: /etc/asterisk/cdr_mysql.conf Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup, SoftHangup
Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR userfield not written into DB
Norbert Zawodsky schrieb: I'm using Asterisk 1.2.12.1 with mysql as the cdr backend. In the dialplan i've written exten = 1234,n,Set(CDR(userfield)=blah) exten = 1234,n,Answer() exten = 1234,n,Queue(.) exten = 1234,n,Hangup() When I'm doing a call I can see that the statement is executed. But when the call finishes, a cdr is written into the DB with an empty 'userfield'. I'm sure, I'm missing something but can't figure out, what... /etc/asterisk/cdr_mysql.conf : [global] userfield=1 ... Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help in installing asterisk
aster...@opensourcesolution.in schrieb: when i am compiling asterisk-1.4.26.3, i am getting errors of dependency. Well, install the dependencies first. :-) What exactly does it complain about? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help in Perl AGI - Bridge 2 channels
velusamy velu schrieb: In Perl AGI, I have two number like 700, 800. I have to call first 700. Next I have to call 800. After that I have to connect this two numbers in the call. How can I do it in Perl AGI? I think you are looking for the Bridge manager command which is available since Asterisk 1.6. ---cut--- Action: Bridge Synopsis: Bridge two channels already in the PBX Privilege: call,all Description: Bridge together two channels already in the PBX Variables: ( Headers marked with * are required ) *Channel1: Channel to Bridge to Channel2 *Channel2: Channel to Bridge to Channel1 Tone: (Yes|No) Play courtesy tone to Channel 2 ---cut--- Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining SIP peer's default mailbox
Steve Johnson schrieb:
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
mailbox as coded in sip.conf.
sip.conf
--
[100]
mailbox=100
[102]
mailbox=102
[103]
mailbox=100
I want a function which I can use in the dialplan (1.6) that works like:
DefaultMailbox(100) - 100
DefaultMailbox(102) - 102
DefaultMailbox(103) - 100
for example:
exten s,n,VoicemailMain(DefaultMailbox(${CALLERID(num)}))
SIPPEER(...|mailbox) I guess.[1] E.g.
VoicemailMain(${SIPPEER(${CALLERID(num)}|mailbox)});
[1] http://www.das-asterisk-buch.de/2.1/functions-sippeer.html
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] How to answer to an incoming call with alsa.
Fabien Comte schrieb: I try to use asterisk as softphone with alsa. I search how to answer to an incoming sip call (from wan). Does anyone did it (extensions.conf exemple) ? Maybe something like Dial(ALSA/hw:0,0); (untested) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Asterisk + Monitor() Poor quality
Danny Nicholas schrieb: WAV49 is by definition lesser quality but Louder sound http://www.vloud.com/ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL problem: bug or feature?
Klaus Darilion schrieb:
forgot to mention this happens on Asterisk 1.4.26.1
Klaus Darilion schrieb:
Hi! I have a problem with jump in AEL:
_+43123456789! = jump +22;
+22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
22 = { NoOp(); }
- OK
_+43123456789! = jump 22;
_22 = { NoOp(); }
- OK
_+43123456789! = jump +22;
_+22 = { NoOp(); }
-- AEL compile error:
LOG: lev:4 file:pbx_ael.c line:1234 func: check_goto Error: file
./ofis/extensions.ael_trunking, line 525-525: goto: no label +22|1
exists in the current context, or any of its inclusions!
Not that it should make a difference (as + is not a special
character in Asterisk's patterns) but did you try
_+43123456789! = jump +22;
_[+]22 = { NoOp(); }
just in case?
Is this is some special feature/limitation or just a bug?
Looks like a bug to me.
Philipp Kempgen
--
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Re: [asterisk-users] OriginateResponse Event
Danny Nicholas schrieb: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? AFAICR ActionID is a string. Probably limited to 255 characters or something. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith On Mon, 2009-10-05 at 14:55 +, Anahi Ludueña wrote: I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Whenever you send an action through AMI, you should also provide an ActionID string, which is something you create and should be unique for each action you send. The response from that action should contain that same ActionID, so that you can identify the responses with the corresponding action based on the ActionID. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
B.Masoud @ SH schrieb: I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel problems on SuSE 9.3
Angus Asterisk schrieb: It seems that the zaptel startup script does not work. I noticed at startup the line: /etc/init.d/zaptel: line 40: /etc/init.d/functions: No such file or directory Line 40: # Source function library. if [ $system = redhat ]; then . $initdir/functions || exit 0 Fi The . %initdir... is line 40. Any ideas how to fix this file on suse? /etc/init.d/functions might be available as /lib/lsb/init-functions so the snippet in /etc/init.d/zaptel could be changed to something like # Source function library. if [ -e /lib/lsb/init-functions ]; then . /lib/lsb/init-functions || exit 0 elif [ -e $initdir/functions ]; then . $initdir/functions || exit 0 fi (untested) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) Zaptel, SuSE 9.3, Debian
Josué Conti schrieb: I had problems with SUSE to Debian that they do not appear, and for the compilation of packages with Debian has always been all that easy, try Debian Etch 4.0, you'll like it. Do you recommend Debian 4 (Etch, oldstable) for a particular reason? Should you have any problems with Debian 5 (Lenny, stable) please report them on the Debian bugtracker. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to remove peers from channels
RSCL Mumbai schrieb: Pls see below output. I would like to remove the last 3 peers. How can I do this ? [trixbox ~]# /usr/sbin/asterisk -rx sip show channels Use grep. (See `man grep`.) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dCAP Exam
Benny Amorsen schrieb: Jared Smith jsm...@digium.com writes: In a nutshell, you can pass the test without having any experience on Polycom IP phones and Digium cards, as long as you know how to use Asterisk itself. I don't think it's unreasonable at all that it is in the test -- if you can't connect SOME kind of phone to Asterisk, you don't deserve certification. They have to pick one brand because it's infeasible to bring 5 different phones for each test taker. What about Bring your own favorite phone? :-) IMHO the Polycoms are a bad choice for the test because they reboot for every modification of the SIP account parameters so unless you have previous experience with the Polycoms you will loose a lot of time. BTW: The default username/password is Polycom/456. The Address field should NOT contain an IP address. Yeah, that's a bit hard to guess. As far as the analog card and DAHDI is concerned: I just skipped this part and didn't even compile DAHDI. Go for the bonus credit: AFAICR it was something about DND and call forwarding. Don't waste your time on ODBC or MySQL or anything that I would normally do. Use AstDB. Make sure you are familiar with either vim or kate/kwrite. Start compiling, and meanwhile read the rest of the test. Don't try to remove the default stuff from the sample config files. Quick dirty is the motto. BTW: The supervisor said he had never seen anybody else use AEL before. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 404 for SUBSCRIBE
yuhu_han schrieb: I always get 404 respond when I send SUBSCRIBE to asterisk. Does anybody know why? Message flow is as follows: SUBSCRIBE sip:1...@192.168.1.32:5060 SIP/2.0 Event: dialog Accept: application/dialog-info+xml SIP/2.0 404 Not Found What does the Asterisk CLI tell you about it at verbosity 3? Show us your dialplan for extension 1006. My guess is that it's lacking the hint priority (core show hints). exten = 1006,hint,SIP/user1006 exten = 1006,1,Dial(SIP/user1006) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday 11th: Aswath Rao: Trapezoidal VoIP is Evil on VoIP Users Conference at Noon EDT
randulo schrieb: We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that Trapezoidal VoIP is Evil. Dean Collins schrieb: It might just be me but what is trapezoidal voip? Here's what I found: http://www.slideshare.net/aswath/trapezoidal-voip-is-evil However OpenID is probably not a solution (as far as I can tell - they slides don't really go into detail). OpenID etc. might be convenient but it's not secure. See http://www.nytimes.com/2008/08/11/technology/11iht-digi11.1.15135411.html http://en.wikipedia.org/wiki/OpenID#Security_and_phishing Forget about OpenID. Use HTTPS and client certificates for single sign-on services. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc insert with mssql
Myles Wakeham schrieb: My biggest negative to ODBC is that the calls that the client application has to make to the database are hard to visualize and debug, particularly if you are crafting a SQL QUERY statement that goes on for pages. Its for this reason that I'd want to have something where I could get more visibility to the DML. It sounds like this might be the exact problem that the original poster was having - that their DML had errors in it, and if it could be visualized before it hit the database for debugging, it might have been quicker to debug. For MySQL you could enable the query log. Don't know if such a thing exists in MsSQL. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan tips
harry R schrieb:
But as someone else wrote before, you can do a dialplan like this
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
exten = 101-BUSY,n,Wait(3)
exten = 101-BUSY,n,VoiceMail(1...@default,b)
exten = 101-BUSY,n,Playback(vm-goodbye)
exten = 101-BUSY,n,Hangup()
exten = _101-.,1,Goto(101-NOANSWER,1)
Such a dialplan is potentially dangerous.
Someone could call 101-BUSY directly and leave a voicemail message
even if you are available.
Please don't jump to extensions. Use labels (priorities) instead.
Or use the switch(){} statement in AEL (:-) so you don't have to
worry about the implementation details.
Philipp Kempgen
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Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Set PHP binary location for AGI
Myles Wakeham schrieb: Steve writes: #!/usr/bin/php -q which I would assume I simply need to change to: #!/usr/local/bin/php -q This should work. Did you try it? Yes, its working fine. if I can rely on the #! setting in the file, that's good enough for me. It's not really a setting and it has nothing to do with Asterisk. It's just an all normal shebang line. http://en.wikipedia.org/wiki/Shebang_%28Unix%29 http://en.wikipedia.org/wiki/Shebang_(Unix) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Elliot Murdock schrieb: I am wondering how the Asterisk community has been working on solutions to deal with the asymmetric quality of ADSL. Voip is becoming popular and a bottleneck does exists on the ADSL upload side. One participant's upload is the other participant's download and vice-versa. So how would different codecs for sending and receiving help? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVITE Privacy Information
Cyprus VoIP schrieb: I ran into this problem: When I change the CALLERID(num and name) to anonymous, they are also changed in the RPID line and not only in the From. OK. I'd try to set sendrpid=no in sip.conf and then add a Remote-Party-ID header in the dialplan. SIPAddHeader(Remote-Party-ID: sip:@...;screen=yes;privacy=full); I have no idea if Asterisk will let you do that even if sendrpid is disabled. Original Message Subject: Re: [asterisk-users] INVITE Privacy Information From: Philipp Kempgen philipp.kemp...@amooma.de To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, 27 July, 2009 17:16:45 Cyprus VoIP schrieb: These are the sip headers I need to add to the INVITE: /P-Asserted-Identity: sip:*1234567...@192.168.1.100:5060 SIPAddHeader(P-Asserted-Identity: sip:@...); // RFC 3325 Remote-Party-ID: sip:*1234567...@192.168.1.100:5060;party=calling;screen=yes;privacy=full Enable sendrpid=yes ; If Remote-Party-ID should be sent in the [general] section in sip.conf. SetCallerPres(prohib_passed_screen); And I need to change the From to /Anonymoussip:anonym...@192.168.1.100/ and to remove the *67 prefix from the INVITE and To lines. Set(CALLERID(num)=anonymous); // RFC 2543 Set(CALLERID(name)=Anonymous); Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reasons to use AEL
Tzafrir Cohen schrieb:
On Sat, Jul 25, 2009 at 01:03:27PM +0200, Philipp Kempgen wrote:
== extensions.conf:
exten = 30,1,Set(x=5)
exten = 30,n,While($[${x} = 9])
exten = 30,n,NoOp(x ist ${x})
exten = 30,n,ExecIf($[${x} 5],ExitWhile)
exten = 30,n,Playback(beep)
exten = 30,n,Set(x=$[${x} + 1])
exten = 30,n,EndWhile()
exten = 30,n,NoOp(done)
== extensions.ael:
30 = {
x=0;
while (${x} = 9) {
NoOp(x ist ${x});
if (${x} 5) {
break;
}
Playback(beep);
y=${x} + 1;
AEL is so easy that you managed to err with it :-p
oops :-)
}
NoOp(done);
}
Philipp Kempgen
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Re: [asterisk-users] INVITE Privacy Information
Cyprus VoIP schrieb: I would like to use Asterisk to add/modify SIP headers in the INVITE message, to include Privacy information, if the INVITE includes a *67 prefix (or another predefined prefix). That's an example of the INVITE I get: /INVITE sip:*6700112233...@192.168.1.100 SIP/2.0 From: 123456789sip:*1234567...@192.168.1.100;tag=3 To: sip:*6700112233...@192.168.1.100 / These are the sip headers I need to add to the INVITE: /P-Asserted-Identity: sip:*1234567...@192.168.1.100:5060 SIPAddHeader(P-Asserted-Identity: sip:@...); // RFC 3325 Remote-Party-ID: sip:*1234567...@192.168.1.100:5060;party=calling;screen=yes;privacy=full Enable sendrpid=yes ; If Remote-Party-ID should be sent in the [general] section in sip.conf. SetCallerPres(prohib_passed_screen); Privacy: id/ SIPAddHeader(Privacy: id); // RFC 3325, RFC 3323 And I need to change the From to /Anonymoussip:anonym...@192.168.1.100/ and to remove the *67 prefix from the INVITE and To lines. Set(CALLERID(num)=anonymous); // RFC 2543 Set(CALLERID(name)=Anonymous); Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disposition answered after authenticate?
Oguzhan Kayhan schrieb:
Problem is, if the user authenticates, * starts counting as billable
seconds even if i hangup the phone before the called party answers..And
also
as disposition.. it accepts all calls authenticated as 'answered'
If i commentout the authentication line everything works as it should be.
How should i use authentication that, it should accept it as aswered by
default.
Here is my dialplan:
[CallingRule_testcall]
exten = _0XX,1,Authenticate(/etc/asterisk/passwords,an)
exten = _0XX,n,Set(CALLERID(num)=0312290${CALLERID(num)})
exten =
_0XX,n,Macro(trunkdial-failover-0.3,${test}/${EXTEN:0},${span_1}/9${EXTEN:0},test,span_1)
Try ResetCDR() after Authenticate().
Philipp Kempgen
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[asterisk-users] Verbose() messages go unnoticed
Does anybody else have the feeling that custom messages
(Verbose(1,...)) do not stand out enough on the CLI?
We're sending messages like Extension 123 is unknown to the output
and that should tell the user why a call to 123 fails but users fre-
quently crank up the verbosity to 3 or 10 so our messages go unnoticed
in many cases.
My idea was to use terminal escape sequences to make my messages
bold and black on yellow background.
apps/app_verbose.c:
static int verbose_exec(struct ast_channel *chan, void *data)
{
...
switch (vsize) {
case 0:
-ast_verbose(%s\n, vtext);
+ast_verbose(\x1B[103;30;1m %s \x1B[0m\n, vtext);
break;
case 1:
-ast_verbose(VERBOSE_PREFIX_1 %s\n, vtext);
+ast_verbose(VERBOSE_PREFIX_1 \x1B[103;30;1m %s \x1B[0m\n,
vtext);
break;
That's just an ugly hack of course but you get the idea.
I have the feeling that whenever somebody uses Verbose() there is a
reason for doing so and thus the messages should be a lot more eye-
catching.
Philipp Kempgen
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Re: [asterisk-users] Verbose() messages go unnoticed
Steve Edwards schrieb: On Sun, 26 Jul 2009, Philipp Kempgen wrote: Does anybody else have the feeling that custom messages (Verbose(1,...)) do not stand out enough on the CLI? Yes. We're sending messages like Extension 123 is unknown to the output and that should tell the user why a call to 123 fails but users frequently crank up the verbosity to 3 or 10 so our messages go unnoticed in many cases. Your users can change the verbosity? Users (== admins) in contrast to developers. My idea was to use terminal escape sequences to make my messages bold and black on yellow background. apps/app_verbose.c: I'm not a big fan of escape sequences and colors because they mung up console log files, greps, text based emails, and not all terminal emulators interpret them correctly even if the TERM environment variable is set correctly. Of course escape sequences should only be used if Asterisk uses them for the normal CLI output as well. The -n option or nocolor=yes in the options section in /etc/asterisk/asterisk.conf should disable them. I think indentation to indicate significance may be better. Currently (1.2), verbose() indents based on an optional level from 0 to 4 as follows: verbose level 0 verbose level 1 == verbose level 2 -- verbose level 3 verbose level 4 verbose level 5 with unspecified messages displayed as level 0 Same thing in version 1.4. If level 0 was indented one more level than 4, your level 1 messages would stick out like the proverbial sore thumb from the fire hydrant of a busy console. Of course, training admins to stop [ab]using noop() would be a great first step :) Well, the AEL compiler adds lots of calls to NoOp(). These priorities serve as jump addresses for control structures like if, switch etc. ... NoOp(Console/dsp, Finish if-if-if-from-pstn-604-606-607) in new stack IMHO these messages are of very little value to the user (== admin) unless he/she was debugging the AEL compiler itself. I'd love if the AEL compiler could downgrade NoOp() to Verbose(6 or higher). But that's a different story. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reasons to use AEL (was: Re: Goto from a feature macro is not working?)
Miguel Molina schrieb:
Philipp Kempgen escribió:
Use macros in AEL so you don't have to care about the underlying
implementation. :-) scnr
Right now for every implementation I made, I didn't have the need to
program in AEL, only plain extensions, some AMI and AGI. But well, it
seems to have a lot of advantages. Please tell me some, I may take a
look to it too see if it's worth spending the time to learn and get the
best out of it.
I'd say control structures (and proper indentation) are one of the
most important reasons to use AEL (conditionals: if .. else, switch
.. case, ..., loops: for, while) because they look so familiar.
Imagine nested control structures in extensions.conf with Goto(),
GotoIf(), While(), EndWhile(), ExitWhile(), ContinueWhile() and
priorities - such code is not what I call maintainable.
== extensions.conf:
exten = 30,1,Set(x=5)
exten = 30,n,While($[${x} = 9])
exten = 30,n,NoOp(x ist ${x})
exten = 30,n,ExecIf($[${x} 5],ExitWhile)
exten = 30,n,Playback(beep)
exten = 30,n,Set(x=$[${x} + 1])
exten = 30,n,EndWhile()
exten = 30,n,NoOp(done)
== extensions.ael:
30 = {
x=0;
while (${x} = 9) {
NoOp(x ist ${x});
if (${x} 5) {
break;
}
Playback(beep);
y=${x} + 1;
}
NoOp(done);
}
In this example, we needed more lines in AEL; if we had added another
command to the if condition in our while loop, ExecIf() would not be
enough anymore and we would be forced to use a more complex construc-
tion with GotoIf(). Our extensions.conf would be a lot longer.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] how to match no callerid in 1.6 ?
Leif Madsen schrieb:
I don't see how using the
exten[/callerid] notation is really better than the GotoIf()
Personally, the GotoIf() makes much more sense to me, because you're placing
the
matching logic in a single place,
True.
as opposed to an error prone method of adding
an ending / at the end of every line of that extension.
exten = _[A-Za-z0-9].,1,Set(EXTENSION=${EXTEN})
exten = _[A-Za-z0-9].,n,Goto(start,1)
exten = start,1,Verbose(2,Incoming call from ${CALLERID(num)} to extension
${EXTENSION})
exten = start,n,GotoIf($[${CALLERID(num)} = 5551212]?bad_callerid,1)
exten = start,n,...
exten = bad_callerid,1,Verbose(2,A very bad man!)
exten = bad_callerid,n,Hangup()
I think that is a better method than constantly typing a complex pattern
match,
BTW: If you hate having to type the same exten = stuff over
and over for each priority: use AEL and let the AEL compiler do the
work. :-)
_[A-Za-z0-9]. = {
Verbose(2,Incoming call from ${CALLERID(num)} to extension ${EXTEN});
if (${CALLERID(num)} = 5551212) {
Verbose(2,A very bad man!);
Hangup();
}
...
}
And while we're at it: In many cases database lookups by means of
DB() or AGI() or a custom ODBC_*() function make even more sense
than a hard-coded list of GotoIf()s resp. if clauses.
Philipp Kempgen
--
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
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Re: [asterisk-users] Reasons to use AEL
Scott Gifford schrieb: Can you recommend a good tutorial or book that covers AEL? http://www.das-asterisk-buch.de/2.1/extensions.ael.html has some examples but unfortunately the explanations are in German. :-) voip-info has some examples as well: http://www.voip-info.org/wiki/view/Asterisk+AEL http://www.voip-info.org/wiki/view/Asterisk+AEL2 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: Regarding using the sip show peers command, I remember somewhere seeing that it only works for static sip accounts and does not list accounts that are dynamically stored in a database. Most of my accounts are database entries, so would the sip show peers command work? Yes it does work, at least with rtcachefriends=yes in sip.conf. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to match no callerid in 1.6 ?
Louis-David Mitterrand schrieb:
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Why remove the elegant and minimal exten/emtpy
notation
Not that need the exten/callerid syntax for anything but I'd say
this is a bug and a regression.
The syntax is exten[/callerid] so the / clearly says that there
is a second argument even if that happens to be an empty string.
Philipp Kempgen
--
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Re: [asterisk-users] Web Browser Pop-up
Vincent Renaville schrieb: An other part of my project is to eneable click-to-call from a web page, do you know a kind of project that implement callto protocol, at this time I use Noojee click but It only work with Firefox. BTW: Sooner or later every major web browser will implement custom URL protocol scheme handlers: http://www.whatwg.org/specs/web-apps/current-work/multipage/browsers.html#custom-handlers Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
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[asterisk-users] notification mail to an user if his vmbox is full (was: Re: dialplan tips)
harry R schrieb: 2) about asterisk voicemail maximum message limit, is it possible to send a notification mail to an user if his vmbox is full ? How can i do that if it's possible. Write a cron job to check if one of the mailboxes is full (ls -l /var/spool/asterisk/voicemail/vmContext/mailBox/INBOX/msg*.txt | wc -l) or use the externnotify parameter in voicemail.conf (3rd arg tells you the number of messages in the INBOX folder) and send an email. You could read the email address from voicemail.conf or from the voicemail Realtime family/table depending on what you use. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto from a feature macro is not working?
Miguel Molina schrieb: I just ran into a similar problem, I needed a macro spreaded over several contexts because it's kind of a part of an IVR. I switched to GoSob() and Return() applications (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub) and everything goes fine now. The only thing to be sure is that you Return() in *every* case from the Macro context(s), to avoid unknown and undesirable results of your calls. Use macros in AEL so you don't have to care about the underlying implementation. :-) scnr Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How determine extension of who initiated call
Michelle Dupuis schrieb:
I'm working on a script that needs to determine the extension (eg: 123) of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
${CALLERID(num)} ?
Philipp Kempgen
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Re: [asterisk-users] CallerPres SIP headers Analog Phone
Maybe https://issues.asterisk.org/view.php?id=71 should be re-opened because the north American vertical service codes are still hard- coded in Zaptel/Dahdi. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Do analog gateways detect a phone is plugged in or out ?
Steve Totaro schrieb: On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood ste...@coppice.orgwrote: Olivier wrote: I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. It certainly would seem possible and would be a great feature request. There probably is no circuitry existing to do it, but I would assume that ohms, volts, or something could be measured while sending a small amount of voltage down the FXS lines. Bonus point will be given for detecting the phone model and color as well. ;-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Function if SIP Device is Still Alive
Elliot Murdock schrieb: I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? You could have a script execute asterisk -rx 'sip show peers' and read the status for each peer. On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold based on user
Juan C. Crespo R. schrieb:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play music
1, if another from Technical Support put on hold the call play music 3,
something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
Some dialplan logic around Set(CHANNEL(musicclass)=...) should do
the trick I guess.
Maybe the easiest way (in Asterisk 1.6) would be to add
setvar=musicclass=admin / setvar=musicclass=support / ...
to your SIP peers and then do something like
_X. = {
Set(CHANNEL(musicclass)=${SIPPEER(${EXTEN},chanvar[musicclass])})
Dial(SIP/${EXTEN});
}
in your dialplan (untested).
Philipp Kempgen
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Re: [asterisk-users] Waiting for a call to complete with AMI Originate
Scott Gifford schrieb: I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or doesn't pick up. The problem I'm having is that when a fax is sent and the line picks up but doesn't accept the fax (for example if I call a voice line). In this case, I don't seem to have enough information to tell when the call has failed and I should give up. I do get a Hangup event, but I don't see a way to distinguish it from other hang-up events from other calls. Here is an example of a recent fax I sent (the format of the request/ response lines is a dump of the variables in Perl, hopefully it makes sense): 'Application' = 'txfax', Any suggestions? Probably not what you are looking for but you could use Iaxmodem + HylaFax. Alternatively have a look at the SendFax() application in Asterisk 1.6. -- -= Info about application 'SendFAX' =- [Synopsis] Send a FAX [Description] SendFAX(filename[|options]): Send a given TIFF file to the channel as a FAX. ... This application sets the following channel variables upon completion: FAXSTATUS - status of operation: SUCCESS | FAILED FAXERROR- Error when FAILED -- Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExecIf and empty variables (early evaluation)
Benny Amorsen schrieb:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
This is fairly harmless in this case, in that it just causes a warning.
You could split it up into multiple statements:
if (${QueueName} != ) {
Set(foo=${QUEUE_WAITING_COUNT(${QueueName})});
} else {
Set(foo=-1); // or whatever
}
(don't remember how to write that in extensions.conf format)
Pros:
- conditional evaluation
- more readable (ExecIf() looks ugly)
Cons:
- more statements
- less readable then a ternary conditional expression in real
programming languages:
foo = ($queuename != ? queue_waiting_count($queuename) : -1)
However, what about this case?
exten = _X!,n,ExecIf($[${bar} 10]?Set(foo=${INC(bar)}))
Which you can argue that this code is in poor taste, it is definitely
surprising that INC is evaluated in this case, changing ${bar} even if
${bar} = 10.
It probably isn't possible to do something about this, but now you have
all been warned... This could be a good reason for avoiding side effects
in functions, and thereby banning ${INC()}
Ban ExecIf(). Use AEL. Use if(){} blocks. :-)
In order to use control structures like if .. else/switch .. case
it's almost necessary to write your dialplan in AEL because the same
thing is so incredibly hard to read and write in extensions.conf
format (GotoIf()).
Although very basic dialplans look good in extensions.conf my
suggestion is to use AEL and let the AEL compiler figure out how to
translate that into an Asterisk dialplan.
Philipp Kempgen
--
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Re: [asterisk-users] CallerPres SIP headers Analog Phone
Ketema Harris schrieb: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? I don't know if/how Asterisk handles/stores CLIR for analog handsets but SetCallerPres(prohib_passed_screen) does the trick when dialing to a SIP channel. Remote-Party-ID: ...;privacy:full;screen:yes You could add a *67 extension to your dialplan and store the CLIR state in AstDB for example. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A reason TO run Asterisk as root
Jeff LaCoursiere schrieb: 2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. That could just as well be a reason NOT to run Asterisk as root. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queues load balancing
Danny Nicholas schrieb:
Here is a brute force solution:
[global]
CALLCOUNT=0
exten = 123,1,Ringing
exten = 123,2,Wait(1)
exten = 123,3,Answer
exten = 123,4,Set(CALLCOUNT)=${CALLCOUNT}+1)
...,Set(CALLCOUNT=$[${CALLCOUNT} + 1])
or
...,Set(CALLCOUNT=${MATH(${CALLCOUNT}+1,int)})
exten = 123,5,Gotoif($[$(CALLCOUNT} = 3]?queue2)
exten = 123,6,Queue(queue_1)
exten = 123,7,Hangup
exten = 123,8(queue2),Set(CALLCOUNT=0)
exten = 123,9,Queue(queue_2)
exten = 123,10,Hangup
Philipp Kempgen
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[asterisk-users] iax.conf, IP-based access control
The documentation in http://svn.digium.com/svn/asterisk/branches/1.4/configs/iax.conf.sample (and http://svn.digium.com/svn/asterisk/branches/1.6.*/configs/iax.conf.sample) seems slightly wrong. --- ; ... Limited IP based ; access control is allowed by use of allow and deny keywords. ... --- allow specifies an allowed codec. It should read: --- ; ... Limited IP based ; access control is allowed by use of permit and deny keywords. ... --- Codecs: disallow/allow Netmasks: deny/permit I think this does not justify filing a bug. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax.conf, IP-based access control
Tilghman Lesher schrieb: On Thursday 16 July 2009 12:26:25 Philipp Kempgen wrote: I think this does not justify filing a bug. No, it does. Go ahead and file it. ok. https://issues.asterisk.org/view.php?id=15518 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
Barry L. Kline schrieb: I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than NOT_INUSE. I have two extensions: and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use to call 6668 and in the dialplan have a noop to see what DEVICE_STATE() is returning for both extensions. What is the super-secret sauce required to get Asterisk to return the correct state? Just to be sure: Do you have hints configured for the extensions? See http://das-asterisk-buch.de/2.1/blf-leds.html (The text is in german but there are many examples in extensions.conf and extensions.ael syntax. Zurück = Previous, Weiter = Next) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.26 final release - What is blocking?
Marc Leurent schrieb: I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. No more issues are targeted for 1.4.26 however I guess if somebody wanted to it wouldn't hurt to test issues in ready for testing state targeted for 1.4.27. https://issues.asterisk.org/view.php?id=14309 https://issues.asterisk.org/view.php?id=15182 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to block inbound call with Asterisk?
VIP Carrier schrieb:
How would you block inbound call's? for example person who is calling me is
212-555-1212, and I would like to do not receive the calls from this person
and give them busy tone.
What should I write in asterisk config files?
core show function CALLERID
Verbose(1,### Inbound call from ${CALLERID(num)});
if (${CALLERID(num)} = 2125551212) {
Verbose(1,### Block this guy);
Busy(5);
Hangup();
}
Dial(...);
and in to witch file should I
write it???
extensions.ael? extensions.conf?
Philipp Kempgen
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Re: [asterisk-users] #exec in #include'd file
Tilghman Lesher schrieb: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Is there a specific reason not to permit #exec in AEL files? It wasn't coded that way, and it's parsed in a completely different way than any other Asterisk configuration file. I don't know the reason Murf didn't do '#exec' specifically, but I suspect it has to do with the complexity thereof. Thanks for the clarification. I was under the (false) assumption that #include and #exec were some kind of preprocessor directives which would be evaluated before any parsing is done, bu that is not true, at least not for extensions.ael. Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. I'm thinking about the options in following: a) load = extconfig ; possible? #exec in extconfig.conf. b) #exec in modules.conf itself Need to figure out if load is enough or if I should preload the module. And that raises the question how often Asterisk will reload extconfig.conf and modules.conf. It certainly reads modules.conf twice on startup and reads extconfig.conf on startup and reload. Do any other events make Asterisk re-read these files? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why CDR is recording dst value = h?
Zeeshan Zakaria schrieb: For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to verify it. Any idea why is this happening and how can I have correct 'dst' value if the caller hangs up first. Maybe endbeforehexten=yes in cdr.conf does what you need. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setvar and transfer (was: Re: How to Change size of CDR(accountcode) variable?)
Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setvar and transfer
Philipp Kempgen schrieb: Benny Amorsen schrieb: Last concern: Does setvar work even for transfers, like accountcode does? I can't answer your question, but transfer != transfer. Some use a feature code in Asterisk, some initiate a transfer on their phone, some use a way to call the Transfer() application. Mixing it up causes a lot of confusion. Many users always call Dial() with the tT args. They want transfers to work, after all. :-) Many of them probably don't need it, because a good number of SIP phones have a Transfer key which initiates a transfer the SIP way. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #exec in #include'd file
Tilghman Lesher schrieb: On Tuesday 14 July 2009 15:35:20 Philipp Kempgen wrote: Tilghman Lesher schrieb: On Monday 13 July 2009 17:19:15 Philipp Kempgen wrote: Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. No, but you can force it by doing an explicit load of a particular module in modules.conf. Explicitly loaded modules are loaded before all automatically-loaded modules. I'm thinking about the options in following: a) load = extconfig ; possible? #exec in extconfig.conf. b) #exec in modules.conf itself It certainly reads modules.conf twice on startup and reads extconfig.conf on startup and reload. Do any other events make Asterisk re-read these files? For modules.conf, no, but extconfig.conf is re-read upon a 'module reload extconfig' command. This permits changing realtime settings without restarting Asterisk. Just found out that even a simple `asterisk -r` causes extconfig.conf to be re-read. That rules extconfig.conf out for what I am trying to do since I don't want to re-generate extensions.ael every time some- body connects to the Asterisk console. Looks like I should go for modules.conf then and implement a mechanism to avoid re-generating extensions.ael on the second pass in modules.conf. E.g. I could make the script not do anything if extensions.ael was modified less than 5 seconds ago. That's not perfect but should do the trick. Alternatively I could use load = pbx_config.so and put the #exec in extensions.conf as murf suggested. Need to play around a bit. Thanks to both of you and sorry for hijacking my own thread a bit. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] #exec in #include'd file
Hi, Is Asterisk supposed to evaluate #exec's in an #include'd file? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #exec in #include'd file
Philipp Kempgen schrieb: Is Asterisk supposed to evaluate #exec's in an #include'd file? I should probably add that it doesn't work for me in case that wasn't obvious. NOTICE[14143]: ael.flex:878 setup_filestack: --Read in included file /e-globals.ael, 1999 chars ERROR[14143]: ael.y:812 ael_yyerror: File: /e-globals.ael, Line 53, Cols: 8-58: Error: syntax error, unexpected word, expecting '=' The line in question is #exec /e-globals.ael.php execincludes is enabled. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #exec in #include'd file
Tilghman Lesher schrieb: On Monday 13 July 2009 01:03:48 pm Philipp Kempgen wrote: Philipp Kempgen schrieb: Is Asterisk supposed to evaluate #exec's in an #include'd file? The directive #exec is not permitted in an AEL configuration file. I see, that would explain why it doesn't work. :-) But in that case it's a documentation issue. The extensions.conf sample says: The #exec command works on all asterisk configuration files. I guess it should read The #exec command works on all asterisk *.conf files except for asterisk.conf. Is there a specific reason not to permit #exec in AEL files? BTW: That's a good example of something to run in /etc/asterisk/startup.d/*.sh. Thread: http://lists.digium.com/pipermail/asterisk-users/2009-May/232318.html http://lists.digium.com/pipermail/asterisk-users/2009-May/232709.html The story is that I already have a script which recursively evaluates #include's and #exec's in AEL files and then writes extensions.ael. I wanted to get rid of the script because there's no clean way to have it run automatically before asterisk is about to be started but now I can't. Is any *.conf file (which permits #exec) guaranteed to be read before extensions.ael? It would then be possible to (ab)use an #exec in there to trigger my generator script (which must not output anything then of course). extconfig.conf? logger.conf? modules.conf? Ugly workaround but doable. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #exec in #include'd file
Danny Nicholas schrieb: Just out of curiousity (haven't got to AEL yet), should these exec's be re-written as AGI calls? I don't think so. #exec's and AGI() are two entirely different concepts. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #exec in #include'd file
Steve Edwards schrieb: On Mon, 13 Jul 2009, Danny Nicholas wrote: Just out of curiousity (haven't got to AEL yet), should these exec's be re-written as AGI calls? An exec is executed when the file is reloaded by Asterisk. Except that in the case of AEL the #exec is not executed at all. :-) An AGI, well, you know :) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] beeping in headsets from queue callers
Darryl Williams schrieb: How do I turn off the beeps in the head sets when customers are waiting in the Queue? ringinuse=no in queues.conf and/or disable call waiting I guess. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate (Executing a System Command)
J. G. schrieb: I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1...@dummy Application: System http://www.voip-info.org/wiki/view/Asterisk+cmd+System Data: /path/to/script I keep getting a Unable to request channel and am not sure what it is looking for in place of Local/1...@dummy. Danny Nicholas schrieb: Why not just Local/1 (unless your server is actually named dummy)? For Local/ channels @... specifies the context, not a peer/hostname. Syntax: Local/extens...@context[/n] Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
Benny Amorsen schrieb: asterisk-us...@rogg.is writes: 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives me what I want. The challenge with AMI is that it is becoming a high-bandwidth channel. If you're only interested in one event type, you will spend quite a bit of CPU time just discarding uninteresting events. Perhaps it would be possible to make events more granular, instead of just on/off? After you're logged in send Action: Events EventMask: call or Action: Events EventMask: call,system if you're interested in reload etc. But you are right. call still gives you events like NewExten, NewChannel etc. apart from ExtensionStatus. NewExten can be pretty verbose. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
Steve Totaro schrieb: It can make 9977.39 Bogocalls of course! Mind to share the formula? Wait. Got it. Bogomips/2. Why on earth isn't that documented?! ;) On Mon, Jul 6, 2009 at 5:17 AM, abdelkaderabdelkader2...@gmail.com wrote: Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 Hardware Information: Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78 PCI Devices none IDE Devices none SCSI Devices -DELL PERC 6/i (Direct-Access) -DP BACKPLANE (Enclosure) -TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices -Dell Computer Corp. -Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub -Dell Computer Corp. Hub Mounted Filesystems: Mount Type Partition Percent Capacity Free Used Size / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB Totals :0% 763.19 GB 1.63 GB 805.54 GB Memory Usage: Type Percent Capacity Free Used Size Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB - Kernel + applications 2% 194.45 MB - Buffers 2% 159.57 MB - Cached 1% 83.21 MB Disk Swap 0% 22.84 GB 0.00 KB 22.84 GB Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel string
Carlos Ruiz Diaz schrieb: When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Doesn't the OriginateResponse give you that information? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (slightly OT) SIP redirect
Does anybody have some pointers on what a SIP user agent should do on receiving a 3xx redirect? RFC 3261 does not go into much detail (to be tried, SHOULD retry). Should the UA prompt the user for confirmation? That seems important since the new address might involve a route with higher cost. http://tools.ietf.org/html/rfc3261#section-13.3.1.2 ---cut--- 13.3.1.2 The INVITE is Redirected If the UAS decides to redirect the call, a 3xx response is sent. ... SHOULD contain a Contact header field containing one or more URIs of new addresses to be tried. ---cut--- http://tools.ietf.org/html/rfc3261#section-21.3.3 ---cut--- 21.3.3 302 Moved Temporarily The requesting client SHOULD retry the request at the new address(es) given by the Contact header field (Section 20.10). The Request-URI of the new request uses the value of the Contact header field in the response. ---cut--- Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
abdelkader schrieb: What is the maximum number of simultaneous calls supported by asterisk. 42. SCNR. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger an action when B number answers the call
selmak se schrieb: Plain text? http://lists.digium.com/pipermail/asterisk-users/2009-July/234274.html Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729a compatibility
Elliot Murdock schrieb: Thank you for that piece of information. Which RFC does it state that the audio name is G729? http://tools.ietf.org/html/rfc3555#section-4.1.9 On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Flemingkpflem...@digium.com wrote: Elliot Murdock wrote: I have a sip device that is sending in the SDP: rtpmap:98 g729a It does not seem like Asterisk is negotiating the codec properly, because while the call rings, the rtp lines fail. However, on other sip devices that have rtpmap:18 g729 in their SDP, things work fine with Digium's commercial g729 license. How do I get 98 g729a recognized by Asterisk? You don't. That's not a standards-compliant way of reporting G.729A in SDP. The RFC says it should be 'G729', but Asterisk also accepts 'G.729' and 'G729A'. It does not accept any lowercase form of the codec name. http://tools.ietf.org/html/rfc3555#section-3 ---cut--- Note that the payload format (encoding) names defined in the RTP Profile are commonly shown in upper case. MIME subtypes are commonly shown in lower case. These names are case-insensitive in both places. ---cut--- Sounds like it should really be case-insensitive but I might easily be mistaken. Didn't dig too deep into RTP/SDP specifications. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding SIP 183 Session Progress handling in Asterisk
Floimair Florian schrieb: I am having trouble with a project concerning the 183 Session Progress SIP messages. Asterisk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. Which version of Asterisk? However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is non-compliant with RFC3261 (SIP). So can someone please tell me the reason, why Asterisk does not support 183 messages without SDP as this would really help me finding arguments in this situation. So far Alcatel just tells us that this is not SIP-compliant and that we have to change things on the Asterisk side, but I'm not quite sure that this is really the case and having arguments could help me clarifying this situation. What they tell you might actually be correct. Not sure. Sip: f.floim...@commend.com file:///T:\KAT\Signaturen\=%22sip:f.floim...@commend.com%22 Something went wrong here. JFYI. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intercepting a Call while ringing a device
Elliot Murdock schrieb: I am looking for a way to dynamically redirect a call while it is ringing to another device. Basically, if a person is far away from his desk, he should have the option to use another phone and pick up the call. Pickup() application? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using http to provision a Grandstrea GXP2000 phone
Julian Lyndon-Smith schrieb: I have a GXP2010 phone, the one with 18 blinky lights ;) I currently provision the phone by writing out the conf file, encoding it and sending it to the tftp server. I was wondering if anyone had managed to automate the web side of provisioning ? Yes. http://www.amooma.de/gemeinschaft/ http://www.amooma.de/gemeinschaft/hilfe/provisioning.html http://www.amooma.de/gemeinschaft/hilfe/community/provisioning-grandstream.html The web site is in German. Sorry. Here's the code: https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/htdocs/prov/grandstream/settings.php Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PHP, AGI, shebang, ? (was: Re: asterisk-users Digest, Vol 59, Issue 62)
Norm Heinen schrieb: On Thu, Jun 25, 2009 at 11:08 AM, Leah Newmarklnewm...@capalon.com wrote: Take a look at this: /var/lib/asterisk/agi-bin/olehphone# head incoming.php displays: #!/usr/bin/php ? Running it shows this: /var/lib/asterisk/agi-bin/olehphone# ./incoming.php #!/usr/bin/php5 -q That's not even the shebang line from the file. Really strange. asterisk-users-requ...@lists.digium.com wrote: From: Danny Nicholas da...@debsinc.com Looking at my ?man php5? ?q is not a valid option. That may be just on Suse. -q is needed if you use php-cgi as the -cli version in order to suppress CGI headers. But even php-cli silently accepts -q even if it's not documented in the man page. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hotdesk and voicemail
Leif Madsen schrieb: Julian Lyndon-Smith wrote: I want to be able to implement hotdesking where an agent will logon to any phone. I got all of that working, without having to reboot phones, but then hit a brick wall. Voicemail. I still want each phone to use the BLF for voicemail indication, and to use the voicemail button to dial voicemail directly. Is it possible to do this dynamically, or will I have to rewrite the phone config and reboot ? With 1.6.2, you can use the MinivmMWI() to enable/disable the MWI for devices as your agents login and logout, and whenever you finish calling Voicemail() or VoicemailMain(). Is it a good idea to tie SIP accounts to the devices instead of to the users? Wouldn't SIP signaling like CCBS/CCNR be tied to the device as well? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
Leah Newmark schrieb: I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx I AGI Rx #!/usr/bin/php5 -q AGI Tx 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping much, and php seems installed perfectly. Other servers running the same AGIs have no such problem. I also have noticed odd behavior. When I edit an AGI, the changes aren't always showing up in the running of the script via asterisk. For example, I tried editing the bash command to read #!/usr/bin/php -q, and got the same response on my agi debug. I know for a fact it's running the script I've edited: Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php and it's not in some other directory. Any input: obvious or not is requested...a few people here are stumped! Are you by any chance mounting two volumes to the same path? Do you use a PHP opcode cache? Did you try to reboot the system? Strange filesystem? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Jabber
jonas kellens schrieb: I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on the configuration of jabber.conf and the integration of jabber asterisk. But I can not find info on how to build jabber-support with Asterisk. I assume you do this via menuselect. So, what about this 'iksemel' ?? On Debian: aptitude search iksemel You might be on your own on other distros: http://code.google.com/p/iksemel/ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
