[asterisk-users] how to know length of file in seconds
Hi List, Is there any way by which we can get the length of any recorded files into seconds ? - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile one way comunication problem
Hy guys, I have setup a chan_mobile trunk on AsteriskNow and Elastix also , using bluetooth libs and asterisk addon module ( asterisk 1.6) using repository , is working fine but only one way , the client is hear me but I can hear.I use Cambrige Silicon Canyon and a Samsung S3310 ,a motherbord gigabyte new. I weird because I setup another computer (Pentium III) and is working no problem is same settings and hardware. Could be because the motherbord is new and blue-libs is old? Thakns -- Eng.Physc .Alexandru Achim National Institute for Lasers, Plasma and Radiation Physics (INFLPR) Solid-State Quantum Electronics Laboratory P.O. Box MG-36, Magurele,Bucharest R-077125 , ROMANIA Phone: +40763.634.348 Email: alexandru.ac...@inflpr.ro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RealTime Queue Logging in 1.8
Thanks for that, the table got created. Should be plain sailing from here on :) Ish On Tue, 2011-06-07 at 10:43 +0530, Satish Barot wrote: I use following for MySQL... CREATE TABLE queue_log( id int(11) NOT NULL auto_increment, time datetime not null, queuename VARCHAR(50), agent VARCHAR(50), callid varchar(32), event VARCHAR(100), data1 VARCHAR(100), data2 VARCHAR(100), data3 VARCHAR(100), data4 VARCHAR(100), data5 VARCHAR(100), PRIMARY KEY (id) ) ENGINE=InnoDB ; Check the link to know the meaning of data1,data2... for Events. http://www.voip-info.org/wiki/view/Asterisk+log+queue_log [SATISH] On Mon, Jun 6, 2011 at 1:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on +MySQL And that table wont create in my database... Thanks Ish Can someone at least point me to the source file that I can analyse to find out the table requirements? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPv6 and IPv4 NAT not working
Hi All, I tried to play a little bit with IPv6 to test our VoIP quality software with IPv6 RTP streams. I add bindaddr=:: to the general section of the sip.conf and netstat shows that Asterisk is listing also on IPv6. My Asterisk server is behind a IPv4 NAT and was working absolutely perfect. But after my bindaddr change I got a problem with external calls. I spend some time to investigate this issue and found out the outbound calls are working. The externaddr is used in the SIP INVITE. If I received a inbound call the externaddr isn't used any more in SDP part of the answer from the Asterisk. The result is one way audio. In addition I saw the following message in the Asterisk log: [May 25 19:18:18] WARNING[3674] chan_sip.c: Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove localnet and/or externaddr settings. I think that is a bug because the externaddr is used correct during the outbound calls. My Asterisk version is 1.8.3.3! Is somebody able to help me with that issue? Should I write a bug ticket for that? best regards Christoph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect intercom to Asterisk?
Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to have something similar by connecting the intercom end in my appartment to a PC running Asterisk that will dial a phone number through SIP. Does someone know if something like that is available ready to use (Arduino, etc.)? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect intercom to Asterisk?
Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to have something similar by connecting the intercom end in my appartment to a PC running Asterisk that will dial a phone number through SIP. Does someone know if something like that is available ready to use (Arduino, etc.)? Thank you. From what I've heard, most people get a standard intercom that connects to a phone system, and plug it into a SIP Adapter like the PAP2T. Never done it myself, so I can't recommend a suitable intercom. Hopefully someone else can. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different callerid for different extensions
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
this dial plan for outbound .But I have some extensions that is like
100-110,200-210,300-310, etc. with this dialplan when I dial from 100
extension callerid will show 044578900 right, and i have 200 extension also
when i dial from this callerid will show 044578900 right. but i need to
difine every extension should be show different callerid . and same as
INbound also.
Please anybody give me short dialplan for this .
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Connect intercom to Asterisk?
On 07/06/11 09:47, Gilles wrote: Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. The company I work for sell SIP door entry phones but I'll not post anything here since this is not a commercial list! We also used to have a DECT door entry system that would do the same thing as that kid has invented a few years ago. It got a small bit on a TV program here in the UK called The Gadget Show. I spent the day with the film crew guys setting up the kit but unfortunately didn't get to meet the rather nice woman who presents the show - they filmed that part another day and edited it all together. Anyway, I don't really see that this is particularly unique, I also know of companies who already make GSM/3G door entry devices too although they are generally aimed at the business market rather than for home use (pricing design reflects this). cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RealTime Queue Logging in 1.8
Realtime table queue_log@asterisk: Column time cannot be a datetime Works fine once the time column is turned to char... On Tue, 2011-06-07 at 10:43 +0530, Satish Barot wrote: I use following for MySQL... CREATE TABLE queue_log( id int(11) NOT NULL auto_increment, time datetime not null, queuename VARCHAR(50), agent VARCHAR(50), callid varchar(32), event VARCHAR(100), data1 VARCHAR(100), data2 VARCHAR(100), data3 VARCHAR(100), data4 VARCHAR(100), data5 VARCHAR(100), PRIMARY KEY (id) ) ENGINE=InnoDB ; Check the link to know the meaning of data1,data2... for Events. http://www.voip-info.org/wiki/view/Asterisk+log+queue_log [SATISH] On Mon, Jun 6, 2011 at 1:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on +MySQL And that table wont create in my database... Thanks Ish Can someone at least point me to the source file that I can analyse to find out the table requirements? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 - subscriptions.
Hi all, I try to figure out why I have empty : sip show subscriptions list in may asterisk 1.6. When device is registering to asterisk I can see in log: NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1010 but sip show subscriptions is just empty. May it be the problem because devices are registering to asterisk from behind NAT? Regards, Jarek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different callerid for different extensions
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
On Tue, Jun 7, 2011 at 2:29 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
this dial plan for outbound .But I have some extensions that is like
100-110,200-210,300-310, etc. with this dialplan when I dial from 100
extension callerid will show 044578900 right, and i have 200 extension also
when i dial from this callerid will show 044578900 right. but i need to
difine every extension should be show different callerid . and same as
INbound also.
Please anybody give me short dialplan for this .
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
--
_
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Re: [asterisk-users] Different callerid for different extensions
Sir,
I have MYsql database in myserver.
On Tue, Jun 7, 2011 at 4:57 PM, Satish Barot satish4aster...@gmail.comwrote:
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
On Tue, Jun 7, 2011 at 2:29 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
this dial plan for outbound .But I have some extensions that is like
100-110,200-210,300-310, etc. with this dialplan when I dial from 100
extension callerid will show 044578900 right, and i have 200 extension also
when i dial from this callerid will show 044578900 right. but i need to
difine every extension should be show different callerid . and same as
INbound also.
Please anybody give me short dialplan for this .
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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asterisk-users mailing list
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--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Refactor of CDR - Comments please.
Hi, Since raising this ticket about broken CDR data: https://issues.asterisk.org/jira/browse/ASTERISK-17826 I have been researching how CDR records work in various circumstances. CEL will do most things that people want, but that does not change that CDR records are likely to persist into future versions, and should be as correct as possible. Is this something people want? Is it worth the risk to change CDRs again? I am working on a patch within 1.6.2, and if it is considered worthwhile I will clean-up and migrate the patch to 1.8 and put it on the review-board It struck me that the fundamental issue was the bridge_cdr construct that appears to have been added in version 1.4. While this solved the problems it was aimed at solving, it appears to have caused other problems, and perhaps even worsened support of CDR records when channels are transferred/masqueraded. The special cdr reset feature seems to be able to clear out valuable CDRs that are no-longer related to the bridge when it ends, and copying bridge_cdr back into chan-cdr often overwrites other valuable CDR data! A classic example of the current failing is: - A UK based receptionist calls Australia. - Then a 2nd call leg is placed UK to America. - The receptionist bridges the 2 calls. The desired outcome would be 2 CDR records detailing the full duration of both calls for billing purposes. Sadly, at present, one of the 2 CDRs stops when the transfer happens because Asterisk sees only one bridge in progress. Here is the outline of my solution: 1) There are 2 basic types of CDR a) A CDR that tracks a running PBX/dialplan. b) A CDR that tracks an outbound dialled channel. 2) Channels can be bridged and re-bridged. When bridging, the existing code provides a way to merge caller (PBX) and callee (Dialed) CDRs, and for consistency this will be maintained as closely as possible. 3) A CDR on a dialled channel should track that channel for its lifetime. 4) A CDR that tracks a running PBX should track that PBX for its lifetime or until merged into a dialled call. Side-note: 3) and 4) Affect the way cdr data is masqueraded. 5) Aim: Valid CDR output should be changed minimally over the current system. 6) Aim: NoCDR, ResetCDR and ForkCDR etc will remain untouched and function the same. 7) A bridge CDR will be un-needed, instead, the bridged CDR will be stored the dialed channel. 8) It is intended that all call legs will have a CDR, which will accurately reflect the duration of that call leg, but no attempt will be made to record transfers/masquerades beyond current mechanisms. CEL data should provide that additional information if needed. The changes to achieve the above are surprisingly minor (relatively speaking). I am testing all of the cases that I can think of: - Simple call in. - Simple call out. - Call-out, then blind transferred by caller and callee. - Call-out, then att. transferred by caller and callee. - AMP or call-file originated call. - Masq-away of Local channel. - Bridge() channels in the dialplan. - Feature park of calls. - Local/ channel calls. - Feature transfer of calls. - Transfer call to IVR/Playback - Transfer IVR/Playback to a handset - Feature Pickup and Pickup() - SIP blind xfer - SIP att xfer - SIP xfer to ringing channel Thoughts? Many thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect intercom to Asterisk?
On Tue, 7 Jun 2011, Gilles wrote: Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. I don't know anything about electronics and would like to have something similar by connecting the intercom end in my appartment to a PC running Asterisk that will dial a phone number through SIP. Does someone know if something like that is available ready to use (Arduino, etc.)? Why bother when you can buy off the shelf stuff to do it for you. e.g. (UK source) http://www.provu.co.uk/door_entry.html It's not exactly rocket science if you already have an asterisk (or other SIP) system that can handle the calls for you. However if he can sell his box for 40 quid, then I'm all for it... Until someone crowbars the whole unit off the wall and steals the SIM card from it and runs up a huge mobile bill... doesn't look like he's thought of that. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different callerid for different extensions
I mean, what do you want to see in callerid when Extensions
100-110,200-210,300-310 dial/receive the calls?
Like, 044578900 for 300, 044578901 for 101 and something like that.
Using,exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
and exten = _0X,6,Set(CALLERID(num)=${outgoing_ident}), will
always set callerid to 044578900 for Extensions 100,200,300; 044578901 for
101,201,301 and so on .
[SATISH]
On Tue, Jun 7, 2011 at 5:11 PM, mahesh katta maheshka...@flexydial.comwrote:
Sir,
I have MYsql database in myserver.
On Tue, Jun 7, 2011 at 4:57 PM, Satish Barot satish4aster...@gmail.comwrote:
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
--
_
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[asterisk-users] Call Conference and Call transfer and Voice mail settings
Hi, what can I do for call tranfer settings and conference settings in asterisk server with SIP extensions, internal and outbound. and also conference if suppose I press the # and then press extenstion no. it will transfer.I tried this in features.conf file . -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different callerid for different extensions
On Tue, Jun 7, 2011 at 5:53 PM, Satish Barot satish4aster...@gmail.comwrote:
I mean, what do you want to see in callerid when Extensions
100-110,200-210,300-310 dial/receive the calls?
Like, 044578900 for 300, 044578901 for 101 and something like that.
Using,exten =
_0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) and exten =
_0X,6,Set(CALLERID(num)=${outgoing_ident}), will always set
callerid to 044578900 for Extensions 100,200,300; 044578901 for 101,201,301
and so on .
suppose I am dialing from 100 extension its show callerid 44578900 , its
sequence , but from 200 extension is dialing same will comming so whenever i
dialing from 200 callerid should be show different like 44578923 it should
be define myself . please give me logical dialplan for this.
[SATISH]
On Tue, Jun 7, 2011 at 5:11 PM, mahesh katta maheshka...@flexydial.comwrote:
Sir,
I have MYsql database in myserver.
On Tue, Jun 7, 2011 at 4:57 PM, Satish Barot
satish4aster...@gmail.comwrote:
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
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Re: [asterisk-users] Connect intercom to Asterisk?
On Tue, 7 Jun 2011 13:06:23 +0100 (BST), Gordon Henderson gordon+aster...@drogon.net wrote: Why bother when you can buy off the shelf stuff to do it for you. The trick is that this connector must work with existing interphones, such as this one at home: http://img220.imageshack.us/img220/8334/intercomhome.jpg So after I add a second pair of wires to the existing intercom, I guess the options are - either an ATA which will connect to the two-wire analog signal and turn it into an SIP end-point, or - simply running a phone cable from the intercom all the way to the Asterisk box where it will be connected to some hardware -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get DTMF in Konference module in Asterisk
Hi Virendra, Set DTMF option in the Makefile to 1 and then recompile/install the app_konference module. Thanks Krishna On Tue, Jun 7, 2011 at 1:31 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I am trying to get DTMF into conference room. for conference I am using Konference module. Konference don't have an option of DTMF gets. Is there any way by which I can get DTMF within conference room? - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trunk between 2 servers
Hello, We have 1 server installed with centos and asterisk and the sip in configured we can do the external and internal call without issue Now I have installed the same centos and asterisk 1.4 in second computer in the same network. My question if there is any possibility to create a trunk between the first and the second PC in oorder do the inbound and outbound calls Thanks and Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 minimum modules/configuration
Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? We have lots of 1.4 boxes in production, and I'm currently setting up a pair of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the existing 1.4 boxes. All the new installs need to do is receive calls via IAX and send them out via SIP to the 1.4 boxes. No ISDN or analogue channels, no voicemail, no conferences, codec translations, etc. - just the minimum number of modules necessary for basic IAX to SIP routing. Suggestions gratefully appreciated, otherwise I guess I'll try disabling everything, then gradually enabling modules as needed :-) Thanks in advance. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trunk between 2 servers
On Tue, 7 Jun 2011, salaheddine elharit wrote: We have 1 server installed with centos and asterisk and the sip in configured we can do the external and internal call without issue Now I have installed the same centos and asterisk 1.4 in second computer in the same network. My question if there is any possibility to create a trunk between the first and the second PC in oorder do the inbound and outbound calls Sure. Depending on your needs, DUNDI may be appropriate. For my needs (1 man band), I just use IAX. SIP will also do the trick, it's just a little bit more complex to set up but should be well within your skill set if you already have SIP configured for [in/out]bound on your first box. After you've configured iax.conf correctly, all you need is something like: exten = *,n,dial(iax2/user:pass@second/exten-in-second-dialplan) In the 'first' box's dialplan. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 minimum modules/configuration
On 06/07/2011 08:04 PM, Chris Bagnall wrote: Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? Basic deployment is hard to specify, but in any case you can use the following modules as a base to build your system. Its a set of modules that provides very basic sip support for asterisk, and it can be considered very close to absolute minimal. You will propably have to add more modules for dialplan apps, channels, codes etc. [modules] autoload=no load = res_musiconhold.so load = res_smdi.so load = res_rtp_asterisk.so load = res_timing_timerfd.so load = codec_ulaw.so load = format_pcm.so load = app_dial.so load = pbx_config.so load = chan_local.so load = chan_sip.so Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium WCTDM24XXP DTMF CallerID
Hello,
I tried using these options in my chan_dahdi.conf, but when i call my
asterisk box using the fxo line, i get this error:
[Jan 9 01:28:03] WARNING[12295]: sig_analog.c:2365 __analog_ss_thread:
DTMFCID timed out waiting for ring. Exiting simple switch
and no callerid appears in my console:
Incoming call from via DAHDI/1-1
i've put this line in my incoming context to show the message above:
Verbose(Incoming call from ${CALLERID(num)} via ${CHANNEL});
Thanks
On Wed, 2011-04-27 at 09:58 -0500, Shaun Ruffell wrote:
On Wed, Apr 27, 2011 at 09:26:24AM -0300, Antonio Modesto wrote:
Good morning,
I have a digium wctdm24xxp in my asterisk box, i am not able to see
the callerid when the call is incoming from the fxo line, i live in
Brazil, how can i change the signaling from fsk to dtmf?
Hello Antonio,
In chan_dahdi.conf you would need to set cidsignalling=dtmf and cidstart=dtmf.
However there may currently be an issue at least with Asterisk 1.6.2 where the
start of the caller ID is not detected. I would be interested to hear what
sort of results you have when you try to set this.
Thanks,
Shaun
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[asterisk-users] why doesn't s accept incoming call
Call from 'sip' to extension '+1xxxyyy' rejected because extension
not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten = s,n(weasels),PlayBack(weasels-eaten-phonesys)
If I set s to _. it works.
Shouldn't s work here? Is it because the extension includes a +?
sean
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Re: [asterisk-users] why doesn't s accept incoming call
sean darcy wrote:
Shouldn't s work here?
S stands for start. Inbound calls via a provider, to your inbound
context would match. For you example, it'd have to be:
[out]
exten = +1xxxyyy,1,NoOp( this is the extension: ${EXTEN})
exten = +1xxxyyy,n,Answer()
exten = +1xxxyyy,n(weasels),PlayBack(weasels-eaten-phonesys)
Doug
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Re: [asterisk-users] why doesn't s accept incoming call
On Tue, 07 Jun 2011 14:17:41 -0400
sean darcy seandar...@gmail.com wrote:
Call from 'sip' to extension '+1xxxyyy' rejected because
extension not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten = s,n(weasels),PlayBack(weasels-eaten-phonesys)
If I set s to _. it works.
Shouldn't s work here? Is it because the extension includes a +?
s is only used when there is no extension.
Using _. poses a potential security risk, if you then use ${EXTEN}
for a subsequent Dial. The best way to do it is to define your
extensions exactly or based on some pattern that doesn't end with ..
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Re: [asterisk-users] why doesn't s accept incoming call
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
sean darcy
Sent: Tuesday, June 07, 2011 2:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] why doesn't s accept incoming call
Call from 'sip' to extension '+1xxxyyy' rejected because
extension
not found in context 'out'.
But
[out]
exten = s,1,NoOp( this is the extension: ${EXTEN})
exten = s,n,Answer()
exten = s,n(weasels),PlayBack(weasels-eaten-phonesys)
If I set s to _. it works.
Shouldn't s work here? Is it because the extension includes a +?
s is NOT a wildcard. s matches empty extension, it does not match all
extensions.
s is seldom used except in macros and with FXO ports (analog or CAS)
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Re: [asterisk-users] Different callerid for different extensions
On 6/7/11 1:59 AM, mahesh katta wrote:
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
you have 9 digits on the starting number 8 digits
on the ending number. i'll assume it's a typo and
the ending number is 04457 (total of 1100 DIDs)
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
this dial plan for outbound .But I have some extensions that is like
100-110,200-210,300-310, etc. with this dialplan when I dial from 100 extension
callerid will show 044578900 right, and i have 200 extension also when i dial
from
this callerid will show 044578900 right. but i need to difine every extension
should be show different callerid . and same as INbound also.
Please anybody give me short dialplan for this .
change prio 2 line to the following:
exten =
_0X,2,Set(outgoing_ident=04457${MATH(8900+${MATH(${CALLERID(num)}-100)})})
i just did this out of my head, i haven't test it.
but this should map all 100-399 extensions to DID 044578900-0444579199
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Re: [asterisk-users] how to know length of file in seconds
On 11-06-07 02:31 AM, virendra bhati wrote: Hi List, Is there any way by which we can get the length of any recorded files into seconds ? $ sox foo.wav -e stat [1] - http://www.thegeekstuff.com/2009/05/sound-exchange-sox-15-examples-to-manipulate-audio-files/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reload chan_dahdi.conf without disconnect active calls
Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reload chan_dahdi.conf without disconnect active calls
Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reload chan_dahdi.conf without disconnect active calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Tuesday, June 07, 2011 5:13 PM To: asterisk-users Subject: [asterisk-users] reload chan_dahdi.conf without disconnect active calls Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option Yes. module reload chan_dahdi.so However, pridialplan= (and a few other options) cannot be changed on a reload. You have to restart Asterisk or unload/load chan_dahdi.so. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trunk between 2 servers
On 6/7/11 7:40 PM, Steve Edwards wrote: Hi! Agree on the IAX. Had never used it, I was a SIP-Only one man band. Not got a VPS in a different location and configured IAX between them. Cool thing is it has encryption built-in, in case you don't have both of the machine locally. As a really ugly fail-over setup I just made a new context in my extentions.conf for iax-in where I also have all my extentions (like all 6! :-). In the sip context in the dialplan I added an entry to dial extention@other_iax node in case of failure. So now I get trunked calls in on one of both boxes (you can setup an SRV to configure the standby machine) and I'll try the ext locally. If that one fails (not registered) it'll send the call over the IAX trunk to the other machine. Fails there: voicemail. Basically there's no end to what you can do this way, like have some trunk operators connected to the machine that's closest by and deliver the calls there yourself over IAX. IAX can have monitoring and encryption, pretty nice stats too. If you managed to setup SIP, IAX is easy. I only used Google basically. Just don't forget the two contexts to prevent loops. And in my ugly work-around you'll have two places to check voicemail, if you would use that. /Robin For my needs (1 man band), I just use IAX. SIP will also do the trick, it's just a little bit more complex to set up but should be well within your skill set if you already have SIP configured for [in/out]bound on your first box. After you've configured iax.conf correctly, all you need is something like: exten = *,n,dial(iax2/user:pass@second/exten-in-second-dialplan) In the 'first' box's dialplan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is wrong in m
Hi everyone,
What is wrong in below asterisk application? The output should be content of
field booth_status from table booths:
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten = _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status FROM
mytable WHERE extension=${CALLERID(num)} ORDER BY id DESC LIMIT 1)
exten = _X.,n,NoOp(allow_call is: ${allow_call})
But I get:
*allow_call is: 4 *while it should actually be ACTIVE or INACTIVE
I want to read the LAST record found under column extension in mytable
and do a NoOp for it's contents. But instead I am getting 4 which I think
refers to the connection ID?!
***There is only one record in my table right now.
Thanks,
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[asterisk-users] PRI hangup request, cause 18
We have 2 PRI from ATT And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause, and clearing call [Jun 7 17:57:33] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/4 got hangup request, cause 18 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong in m
On Tue, 7 Jun 2011, Bruce B wrote:
What is wrong in below asterisk application? The output should be
content of field booth_status from table booths:
I don't see 'booth_status' or 'booths' anywhere below.
[extension-status]
exten = _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten = _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status FROM
mytable WHERE extension=${CALLERID(num)} ORDER BY id DESC LIMIT 1)
exten = _X.,n,NoOp(allow_call is: ${allow_call})
0) The verbose() application is a 'better practice' than relying on the
obtuse side effect of noop().
1) If you execute the above snippet, what shows on the console log?
2) If you snip the select statement from the console log and paste it into
the mysql application's command line interface, what do you get?
I've never used the dialplan MySQL interface so I may be missing something
obvious. (I prefer to do database activities in AGIs.)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] reload chan_dahdi.conf without disconnect active calls
hi: there is no way to do that. why do you do that? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 21:12:37 + Subject: [asterisk-users] reload chan_dahdi.conf without disconnect active calls Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI issue its BUSY
hi: make sure your pri is up and active. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I use phone line to recive faxes?
hi: yes, make sure you also have a fxs to connect your fax if you want to receive fax by fax Mac. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Fri, 3 Jun 2011 09:30:49 +0100 Subject: Re: [asterisk-users] Can I use phone line to recive faxes? On Thursday 02 Jun 2011, khalid touati wrote: Hi Guys, Actually My question is as in the subject, may I use a regular phone line to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8. Yes, you can. BUT, you will need some sort of FXO interface (allows the computer to connect to the telephone socket on the wall), which is supported by DAHDI (or its predecesor, Zaptel). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is wrong in m
The problem is the OP never performs a Fetch of the data returned by the
Query...
From the VoIP-info page for Cmd MYSQL
MYSQL(Query resultid ${connid} query-string)
Executes standard MySQL query contained in query-string using established
connection identified by ${connid}. Result of query is stored in ${resultid}.
MYSQL(Fetch fetchid ${resultid} var1\ var2\ ...\ varN)
If any rows are available to select, ${fetchid} is set to 1 and a single row is
fetched from a result set contained in ${resultid}. The return fields are
assigned to ${var1}, ${var2} ... ${varN} respectively. If no rows are left to
select, ${fetchid} is set to 0 and ${var1}, ${var2} ... ${varN} remain
unchanged. --
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[asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.
Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: j...@slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
