Re: [asterisk-users] How can I Disable call-waiting
reza naraghi wrote: Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you It is known, as a matter of established fact, that it is possible to disable call waiting on the eyeBeam phone. How to do it is not something in which I can instruct you, but I gather it's a fairly straightforward process, especially if you are autoprovisioning via the textual configuration file. I do not see why you can't enable a call-limit of 1 on the SIP peers for the phones. That doesn't translate to a call-limit on your inbound trunks. It will just cause the call to fail. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I Disable call-waiting
On Wed, 23 Jul 2008, reza naraghi wrote: Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. Try reading your phones manual. This is a phone function, not asterisk. Which phone? People here might be able to help you, even though it's not asterisk specific... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Hi Mark, I assure my queues.conf is full of autopause = no, in the singles and general contexts (I'm not sure where to put it 'cause I found no docs about it). Moreover, this morning I checked my Asterisk with show queues and I found another surprise: SIP/17 with penalty 1 (paused) (Not in use) has taken no calls yet SIP/50 with penalty 1 (paused) (Unavailable) has taken no calls yet How can an unavailable phone (it is not connected on LAN) be paused??? So I wonder...what is the rule that makes a phone paused, then? Another thing I do not understand...when I restart Asterisk, my bunch of disconnected phones have different statusIAX phones are marked with Invalid while SIP are marked with Unavailable...why? What's the difference? The mystery goes on Ah..I forgot to say I do not use agents but only static queues, no real time stuff. Giorgio Mark Michelson wrote: Giorgio Incantalupo wrote: Hi Mark, it is show queues I use to see if phones are paused or not. The phones I'm using for tests are all SIP phones. Yes, what you are supposing could be right...Asterisk could see the phones as stuck. I'm still investigating, making test on my 1.4 box and I have noticed some other strange things about the phones. Some phones when normally used (I made a test making an outbound call) are seen as paused (In use) while other are marked as In Use only: (from Asterisk CLI): SIP/8 with penalty 1 (In use) has taken 1 calls (last was 3247 secs ago)(my phone) SIP/36 with penalty 1 (paused) (In use) has taken no calls yet(my test phone) The phones are the same model and have same sip.conf definition. The queues.conf definitions are the same for the two queues the phones are in. I do not know why queues show shows paused or not for similar phones. Can this be useful!?!? Giorgio The only way that a phone should become automatically paused is if the autopause option is set in queues.conf for the queue. There are ways through the dialplan and manager to manually pause a queue member, but there are no other ways for a member to become automatically paused. That being said, it could be that you have discovered some sort of bug in 1.4. When does this appear to happen? Does it happen randomly or is the situation reproduceable? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.
Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o [LD] res_adsi.o - res_adsi.so [CC] res_agi.c - res_agi.o [LD] res_agi.o - res_agi.so [CC] res_clioriginate.c - res_clioriginate.o [LD] res_clioriginate.o - res_clioriginate.so [CC] res_convert.c - res_convert.o [LD] res_convert.o - res_convert.so [CC] res_crypto.c - res_crypto.o [LD] res_crypto.o - res_crypto.so collect2: ld terminated with signal 11 [Segmentation fault] /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2363 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2365 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2366 make[1]: *** [res_crypto.so] Error 1 make: *** [res] Error 2 -- Debian 4.0 with the latest updates uname -a: Linux vs1201 2.6.18 #2 SMP Tue Oct 23 22:39:08 CEST 2007 x86_64 GNU/Linux ld: GNU ld version 2.17 Debian GNU/Linux gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Any Idea why? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.
On Wed, Jul 23, 2008 at 10:12:21AM +0200, Carsten Bock wrote: Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o [LD] res_adsi.o - res_adsi.so [CC] res_agi.c - res_agi.o [LD] res_agi.o - res_agi.so [CC] res_clioriginate.c - res_clioriginate.o [LD] res_clioriginate.o - res_clioriginate.so [CC] res_convert.c - res_convert.o [LD] res_convert.o - res_convert.so [CC] res_crypto.c - res_crypto.o [LD] res_crypto.o - res_crypto.so collect2: ld terminated with signal 11 [Segmentation fault] /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2363 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2365 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2366 make[1]: *** [res_crypto.so] Error 1 make: *** [res] Error 2 -- Debian 4.0 with the latest updates uname -a: Linux vs1201 2.6.18 #2 SMP Tue Oct 23 22:39:08 CEST 2007 x86_64 GNU/Linux ld: GNU ld version 2.17 Debian GNU/Linux gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Any Idea why? No, but in order to get some clues: Do you still get a segfault by re-running 'make'? To get the exact build command: make NOISY_BUILD=yes -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Overlap dialing via SIP
On Mon, Jul 21, 2008 at 05:10:15PM +0100, Ben Thompson wrote:
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial
${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial
${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()
exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()
This works fairly well but I have noticed that occasionally the WaitExten
feature does
not seem to catch the first digits if they are dialed too quickly. It is
almost as if
there is a some sort of delay and the thirteenth digit is sometimes missed.
In answer to my own email I have found that the Background() function
works slightly better :-
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Background()
exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial ${oldexten}${enddigits})
exten = _X,3,Goto(international-dialout,${oldexten}${enddigits},1)
exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial ${oldexten}${enddigits})
exten = _XX,3,Goto(international-dialout,${oldexten}${enddigits},1)
exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial
${oldexten}${enddigits})
exten = _XXX,3,Goto(international-dialout,${oldexten}${enddigits},1)
exten = t,1,NoOp(timeout so dial just 12 digits ${oldexten})
exten = t,2,Goto(international-dialout,${oldexten}${enddigits},1)
[international-dialout]
exten = _900XX,1,Macro(dialout-pstn)
exten = _900XXX,1,Macro(dialout-pstn)
exten = _900,1,Macro(dialout-pstn)
exten = _900X,1,Macro(dialout-pstn)
In general I have found that Overlap Dialing works very well and it is
a worthwhile feature to have. If there are any others in the UK
who would like to collaborate with me on maintaining an up to date list
of UK mappings please let me know. I would be happy to maintain a
webpage or somthing like that where people could access the info in an
asterisk friendly format.
Ben Thompson
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Re: [asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.
Thanks for the hint with make NOISY_BUILD=yes: In main/db1-ast/hash/hash_page.c Line 654, function first_free(map) there was an error at mask = mask 1; hash/hash_page.c:659: error: stray '`' in program I'm currently recompiling it from the start again, to test it. Tzafrir Cohen schrieb: On Wed, Jul 23, 2008 at 10:12:21AM +0200, Carsten Bock wrote: Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o [LD] res_adsi.o - res_adsi.so [CC] res_agi.c - res_agi.o [LD] res_agi.o - res_agi.so [CC] res_clioriginate.c - res_clioriginate.o [LD] res_clioriginate.o - res_clioriginate.so [CC] res_convert.c - res_convert.o [LD] res_convert.o - res_convert.so [CC] res_crypto.c - res_crypto.o [LD] res_crypto.o - res_crypto.so collect2: ld terminated with signal 11 [Segmentation fault] /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2363 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2365 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2366 make[1]: *** [res_crypto.so] Error 1 make: *** [res] Error 2 -- Debian 4.0 with the latest updates uname -a: Linux vs1201 2.6.18 #2 SMP Tue Oct 23 22:39:08 CEST 2007 x86_64 GNU/Linux ld: GNU ld version 2.17 Debian GNU/Linux gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Any Idea why? No, but in order to get some clues: Do you still get a segfault by re-running 'make'? To get the exact build command: make NOISY_BUILD=yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I Disable call-waiting
Alex Balashov wrote: It is known, as a matter of established fact, that it is possible to disable call waiting on the eyeBeam phone. How to do it is not something in which I can instruct you, but I gather it's a fairly straightforward process, especially if you are autoprovisioning via the textual configuration file. Since when does eyeBeam have any kind of autoprovisioning? I've not seen any reference to it in the manual or on their web site and I /have/ gone looking for it. If I've missed something, I'd be extremely grateful if you could point it out - this is a feature I've wanted for a /long/ time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.2: Linking res_crypto causes segmentation fault.
I just re-unpacked asterisk-1.4.21.2.tar.gz and there was no '`' in the function mentioned below. I have no idea where it came from, i didn't edited the file before. May be something is (terrible) wrong with the server i installed it on ... :) = now it compiles and works Carsten Bock schrieb: Thanks for the hint with make NOISY_BUILD=yes: In main/db1-ast/hash/hash_page.c Line 654, function first_free(map) there was an error at mask = mask 1; hash/hash_page.c:659: error: stray '`' in program I'm currently recompiling it from the start again, to test it. Tzafrir Cohen schrieb: On Wed, Jul 23, 2008 at 10:12:21AM +0200, Carsten Bock wrote: Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: -- [CC] res_adsi.c - res_adsi.o [LD] res_adsi.o - res_adsi.so [CC] res_agi.c - res_agi.o [LD] res_agi.o - res_agi.so [CC] res_clioriginate.c - res_clioriginate.o [LD] res_clioriginate.o - res_clioriginate.so [CC] res_convert.c - res_convert.o [LD] res_convert.o - res_convert.so [CC] res_crypto.c - res_crypto.o [LD] res_crypto.o - res_crypto.so collect2: ld terminated with signal 11 [Segmentation fault] /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2363 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2365 /usr/bin/ld: BFD 2.17 Debian GNU/Linux assertion fail ../../bfd/elflink.c:2366 make[1]: *** [res_crypto.so] Error 1 make: *** [res] Error 2 -- Debian 4.0 with the latest updates uname -a: Linux vs1201 2.6.18 #2 SMP Tue Oct 23 22:39:08 CEST 2007 x86_64 GNU/Linux ld: GNU ld version 2.17 Debian GNU/Linux gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Any Idea why? No, but in order to get some clues: Do you still get a segfault by re-running 'make'? To get the exact build command: make NOISY_BUILD=yes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I Disable call-waiting
Rob Hillis wrote: Since when does eyeBeam have any kind of autoprovisioning? I've not seen any reference to it in the manual or on their web site and I /have/ gone looking for it. If I've missed something, I'd be extremely grateful if you could point it out - this is a feature I've wanted for a /long/ time. To answer my own question, CounterPath have made the somewhat questionable decision to only provide provisioning via HTTP/HTTPS to it's non-retail customers - i.e. if you buy a minimum of hundreds of licences. I now somewhat remember being considerably irritated by this some time ago. CounterPath would not respond to my questions as to why this feature had been removed for retail customers. If there were another usable softphone not tied to a specific platform (such as Cisco CallManager) that had proper support for Plantronics CS60 USB headsets, I would have made the switch ages ago. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] next priority from Dial in Asterisk 1.6
Hello, I'm testing Asterisk 1.6 (from SVN). In my dialplan I have: -- exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,2,Verbose(After Dial) -- If IP denies the call I receive: == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/401113-08200990, SIP/[EMAIL PROTECTED],,tTwWg) in new stack == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- Got SIP response 484 Address Incomplete back from 212.121.243.35 == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (usuarios, 004477, 1) exited INCOMPLETE on 'SIP/401113-08200990' Why is not executing the Verbose after the Dial? Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. It looks like someone at bugs.digium has found what it was, so a fix should be coming soon. PaulH I guess that since there was no mention of this fix in 1.4.21.2 that it's still an open issue? Can you reference the bud ID at Digium so I can follow along? I didn't see it, but might not have known what I was looking for. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: How can I Disable call-waiting
Have you tried incominglimit=1 on sip.conf ?? It worked for me, no matter which softphone or ipphone / ATA I use, it works. You have to use it inside the configuration for every sip peer, just like this: [1002] Type=friend Host = dynamic Port = 5060 incominglimit=1 . . . De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de reza naraghi Enviada em: quarta-feira, 23 de julho de 2008 02:56 Para: asterisk-users@lists.digium.com Assunto: [***SPAM*** Score/Req: 10.0/5.0] [asterisk-users] How can I Disable call-waiting Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you Best regards -- Naraghi e-mail1:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Visioncom Tecnologia da Informacao (www.visioncom.com.br) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with asterisk 1.4.21.1 and h323
Hi to all, i'm experiencing a problem with an h323 trunk between a
Cisco Callmanager 4.2.
I'm using asterisk 1.4.21.1, openh323_v1_18_0, pwlib_v1_10_0
The problem is that sometimes (1 call every 20... but sometimes often)
the call arrives correctly on Call Manager side, and when is answered
after 1-2 seconds Asterisk gives a service unavailable error.
I've noticed enabling h323 trace that when the call is rejectedi i've
got an empty capabilityTable in trace.
When the call works i have:
capabilityTable = 10 entries {
[0]={I
capabilityTableEntryNumber = 1
capability = receiveAudioCapability g7231 {
maxAl_sduAudioFrames = 1
silenceSuppression = TRUE
}CLI
}1*CLI
[1]={I
capabilityTableEntryNumber = 2
capability = receiveAudioCapability g7231 {
maxAl_sduAudioFrames = 1
silenceSuppression = FALSE
}CLI
}1*CLI
[2]={I
capabilityTableEntryNumber = 3
capability = receiveAudioCapability gsmFullRate {
audioUnitSize = 33
comfortNoise = FALSE
scrambled = FALSE
}CLI
}1*CLI
[3]={I
capabilityTableEntryNumber = 4
capability = receiveAudioCapability g711Ulaw64k 20
}1*CLI
[4]={I
capabilityTableEntryNumber = 5
capability = receiveAudioCapability g711Alaw64k 20
}1*CLI
[5]={I
capabilityTableEntryNumber = 6
capability = receiveAudioCapability g729AnnexA 2
}1*CLI
[6]={I
capabilityTableEntryNumber = 7
capability = receiveAudioCapability g729 2
}1*CLI
[7]={I
capabilityTableEntryNumber = 8
capability = receiveUserInputCapability hookflash null
}1*CLI
[8]={I
capabilityTableEntryNumber = 9
capability = receiveRTPAudioTelephonyEventCapability {
dynamicRTPPayloadType = 101
audioTelephoneEvent = 0-16
}CLI
}1*CLI
[9]={I
capabilityTableEntryNumber = 10
capability = receiveUserInputCapability dtmf null
}1*CLI
}k01*CLI
When the call doesn't works i haven't any capabilityTable in trace.
How can i fix that?
My h323.conf is very simple:
[general]
port = 1720
bindaddr = 192.168.1.1
allow=all
tunneling=cisco
[ccm01]
type=peer
host=192.168.1.2
fastStart=no
Thanks to all in advance
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
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Re: [asterisk-users] RES: How can I Disable call-waiting
Hello thank u for ur attention but I did it and in fact its the same as call-limit in newer versions. this cmd limit ur call not disable call-waiting. best regards On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cordeiro [EMAIL PROTECTED] wrote: Have you tried incominglimit=1 on sip.conf ?? It worked for me, no matter which softphone or ipphone / ATA I use, it works. You have to use it inside the configuration for every sip peer, just like this: [1002] Type=friend Host = dynamic Port = 5060 incominglimit=1 . . . *De:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *Em nome de *reza naraghi *Enviada em:* quarta-feira, 23 de julho de 2008 02:56 *Para:* asterisk-users@lists.digium.com *Assunto:* [***SPAM*** Score/Req: 10.0/5.0] [asterisk-users] How can I Disable call-waiting Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you Best regards -- Naraghi e-mail1:[EMAIL PROTECTED] [EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] [EMAIL PROTECTED] Visioncom Tecnologia da Informacao (www.visioncom.com.br) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Reza Jokar Naraghi Tel : (+98)2177360257 Fax : (+98)2177063408 Cell : (+98)9126970085 Cell2:(+98)9366997249 website : www.cac.ir e-mail1:[EMAIL PROTECTED] [EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4
Giorgio Incantalupo wrote: Hi Mark, I assure my queues.conf is full of autopause = no, in the singles and general contexts (I'm not sure where to put it 'cause I found no docs about it). Moreover, this morning I checked my Asterisk with show queues and I found another surprise: SIP/17 with penalty 1 (paused) (Not in use) has taken no calls yet SIP/50 with penalty 1 (paused) (Unavailable) has taken no calls yet How can an unavailable phone (it is not connected on LAN) be paused??? So I wonder...what is the rule that makes a phone paused, then? The paused logic resides fully within the Queue application. For static members, the only way to pause is if autopause is enabled or if the member is manually paused either through the dialplan or manager. Autopause takes effect whenever the queue attempts to ring a member and is unsuccessful. Since you have autopause=no in your queues.conf file, then autopause should not occur on the phones at all. In a further effort to debug the problem, you can check both your console logs and the queue_log to see if there are any messages about the members becoming paused. By the way, I don't think it's come up yet, but which version of 1.4 are you using? If you're not using the latest release, it may be worth it to try using it to see if the same behavior occurs. Another thing I do not understand...when I restart Asterisk, my bunch of disconnected phones have different statusIAX phones are marked with Invalid while SIP are marked with Unavailable...why? What's the difference? The mystery goes on The status reported comes from the device state subsystem. Regarding the IAX channels being marked Invalid, this most likely comes from the fact that app_queue.so is being loaded before chan_iax2.so, meaning that at the time that app_queue checks the device state of those IAX channels, the channel driver has not loaded and so the device state system reports those channels as Invalid. When the phones undergo some state change, or if you issue a module reload chan_iax2.so when the phones are Invalid they will most likely change to the proper state. A better solution is to edit modules.conf to force app_queue.so to load after chan_iax2.so. The SIP phones reporting Unavailable happens most likely because you have a qualify setting in sip.conf, which causes the phones to be Unavailable until qualify determines that the phone is available. Ah..I forgot to say I do not use agents but only static queues, no real time stuff. Giorgio Mark Michelson Mark Michelson wrote: Giorgio Incantalupo wrote: Hi Mark, it is show queues I use to see if phones are paused or not. The phones I'm using for tests are all SIP phones. Yes, what you are supposing could be right...Asterisk could see the phones as stuck. I'm still investigating, making test on my 1.4 box and I have noticed some other strange things about the phones. Some phones when normally used (I made a test making an outbound call) are seen as paused (In use) while other are marked as In Use only: (from Asterisk CLI): SIP/8 with penalty 1 (In use) has taken 1 calls (last was 3247 secs ago)(my phone) SIP/36 with penalty 1 (paused) (In use) has taken no calls yet(my test phone) The phones are the same model and have same sip.conf definition. The queues.conf definitions are the same for the two queues the phones are in. I do not know why queues show shows paused or not for similar phones. Can this be useful!?!? Giorgio The only way that a phone should become automatically paused is if the autopause option is set in queues.conf for the queue. There are ways through the dialplan and manager to manually pause a queue member, but there are no other ways for a member to become automatically paused. That being said, it could be that you have discovered some sort of bug in 1.4. When does this appear to happen? Does it happen randomly or is the situation reproduceable? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
David Nedved wrote: I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. It looks like someone at bugs.digium has found what it was, so a fix should be coming soon. PaulH I guess that since there was no mention of this fix in 1.4.21.2 that it's still an open issue? Can you reference the bud ID at Digium so I can follow along? I didn't see it, but might not have known what I was looking for. The only changes provided in 1.4.21.2 are the two IAX2 security vulnerability fixes mentioned in AST-2008-010 and AST-2008-011. I believe the bug that you want is here: http://bugs.digium.com/view.php?id=12921 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3-way calling for IAX channels
We are developing an softphone based on IAX client version 1.2 (my current SIP softphone has many eoors), but it doesn´t have a specific function for Conferencing (3-way calling) or to place the other party on HOLD. I´m trying to do it through the PBX because our softphone´s lack of functions. I´ll be gratefull for further comments. Thanks again, Daniel On Tue, Jul 22, 2008 at 11:49 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I Disable call-waiting
Hello are you using FreePBX for your configurations? there is an option in the extentions.conf for queues called CWIGNORE=TRUE try disabling it and see if it works for you .. this is the best i can help you with .. i am using call-limit combined with busy-limit to stop the call waiting.. i can't test of a live business server so test it and let me know.. regards Tarek Sawah IT Development Advisor Integrated Digital Systems +963944618286 Date: Wed, 23 Jul 2008 09:25:55 +0330 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] How can I Disable call-waiting Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you Best regards -- Naraghi e-mail1:[EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] _ Time for vacation? WIN what you need- enter now! http://www.gowindowslive.com/summergiveaway/?ocid=tag_jlyhm___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: RES: How can I Disable call-waiting
Ok, I just tested and it works as I said before, here is the log of the second call trying to come in: -- Executing [EMAIL PROTECTED]:2] Dial(DGV/32, SIP/1001|20|tT) in new stack [Jul 23 12:05:55] ERROR[428]: chan_sip.c:3057 update_call_counter: Call to peer '1001' rejected due to usage limit of 1 -- Couldn't call 1001 == Everyone is busy/congested at this time (0:0/0/0) -- Executing [EMAIL PROTECTED]:3] VoiceMail(DGV/32, [EMAIL PROTECTED]) in new stack As the second call came in, I didn't hear the call-waiting beep and the caller of the second call was redirected to the mailbox as my dialplan is setup to do. I hope it helps. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de reza naraghi Enviada em: quarta-feira, 23 de julho de 2008 10:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: How can I Disable call-waiting Hello thank u for ur attention but I did it and in fact its the same as call-limit in newer versions. this cmd limit ur call not disable call-waiting. best regards On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cordeiro [EMAIL PROTECTED] wrote: Have you tried incominglimit=1 on sip.conf ?? It worked for me, no matter which softphone or ipphone / ATA I use, it works. You have to use it inside the configuration for every sip peer, just like this: [1002] Type=friend Host = dynamic Port = 5060 incominglimit=1 . . . De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de reza naraghi Enviada em: quarta-feira, 23 de julho de 2008 02:56 Para: asterisk-users@lists.digium.com Assunto: [***SPAM*** Score/Req: 10.0/5.0] [asterisk-users] How can I Disable call-waiting Hello I really need to disable the call-waiting on my sip phones I studied most of the posts on internet and did it on my asterisk but not useful. in fact I need a comment that I disable call-waiting but without enable call-limit because I want to keep the waited caller on a queue. I tried many states on sip.conf and also users.conf but I didn't do any changes on my extensions.conf and I don't know am I right? If you can help me to hear the busy tone(!!) on my eyebeam sip phone, in fact you return my work to me. thank you Best regards -- Naraghi e-mail1:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Visioncom Tecnologia da Informacao (www.visioncom.com.br) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Reza Jokar Naraghi Tel : (+98)2177360257 Fax : (+98)2177063408 Cell : (+98)9126970085 Cell2:(+98)9366997249 website : www.cac.ir e-mail1:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] e-mail2:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Visioncom Tecnologia da Informacao (www.visioncom.com.br) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
Hi, On Jul/23/2008, Carles Pina i Estany wrote: I'm testing Asterisk 1.6 (from SVN). In my dialplan I have: -- exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,2,Verbose(After Dial) -- Also this doesn't work either: exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,n,Verbose(After Dial) I mean, like before, after some SIP responses like 484 is not executing the after dialing command. In Asterisk 1.4.21.1 it was working as I expected. Is it a feature in Asterisk 1.6? or a bug? After 404 it's going to next priority, but not after 484. Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3-way calling for IAX channels
Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved and that he would have figured out, if he had bothered to try the latest release. 1.4.21.1 should have been fixed, as well, but at that point, I had just spent 3 hours working frantically to get two security advisories out the door, so that the community wouldn't be vulnerable to two critical issues, and suggesting that he try a version that was vulnerable would have been bad. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote: On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved and that he would have figured out, if he had bothered to try the latest release. 1.4.21.1 should have been fixed, as well, but at that point, I had just spent 3 hours working frantically to get two security advisories out the door, so that the community wouldn't be vulnerable to two critical issues, and suggesting that he try a version that was vulnerable would have been bad. Oh, sure. I'm just sayin... It's pretty clear to me that while you're playing in the NFL, he may not be. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
Tilghman Lesher schrieb: On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved While it may sound rude that's absolutely correct. As a software developer in many cases you are more or less sure that an issue has already been solved so you expect the user to upgrade to the latest version or at least to the latest minor version. Having to hunt down problems in old versions is annoying especially for issues that have probably already been addressed. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suddenly my Asterisk Box Hanged up all calls
I´ll be upgrading my box this weekend and let you know the consequences. I´m new at the community and it would be good for me to know what was the problem with 1.4.17 Thanks for taking some time for me. Daniel On Wed, Jul 23, 2008 at 10:59 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote: On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote: On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote: On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote: My * version: 1.4.17 Please upgrade to 1.4.21.2. Just a suggestion, Tilghman: it might have been nice to add because it fixes your specific problem, so that we wouldn't assume because we don't want to talk to you if you rev is too old. :-) It probably fixes his specific problem, AND because I don't like diagnosing an issue that we've already solved and that he would have figured out, if he had bothered to try the latest release. 1.4.21.1 should have been fixed, as well, but at that point, I had just spent 3 hours working frantically to get two security advisories out the door, so that the community wouldn't be vulnerable to two critical issues, and suggesting that he try a version that was vulnerable would have been bad. Oh, sure. I'm just sayin... It's pretty clear to me that while you're playing in the NFL, he may not be. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
Carles Pina i Estany wrote:
Hi,
On Jul/23/2008, Carles Pina i Estany wrote:
I'm testing Asterisk 1.6 (from SVN). In my dialplan I have:
--
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,2,Verbose(After Dial)
--
Also this doesn't work either:
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,n,Verbose(After Dial)
I mean, like before, after some SIP responses like 484 is not executing
the after dialing command.
In Asterisk 1.4.21.1 it was working as I expected.
Is it a feature in Asterisk 1.6? or a bug?
After 404 it's going to next priority, but not after 484.
Thanks,
--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] 3-way calling for IAX channels
2008/7/23 MFH [EMAIL PROTECTED]: Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. I believe that response is slightly misleading - Asterisk does not support conferencing without using meetme, but Zaptel/DAHDI will emulate the PSTN flash/recall facility which looks a bit like a conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI channel types, the endpoint must manage the equivalent of a PSTN flash/recall conference. Anything cross-channel or otherwise more complex does indeed require app_meetme. Given that the OP was referring to IAX, I believe they will need app_meetme. Of course I could be wrong :) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
Hi,
On Jul/23/2008, Anthony Francis wrote:
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
when I said IP i meant the IP value :-) not the two chars string IP.
Sorry for the confusion.
I would shoot my foot not! On Asterisk 1.6, if Dial fails the dialplan
goes to the next one (or n+101, etc.)
In Asterisk 1.6 it tries to go to the invalid extension:
[Jul 23 19:16:54] WARNING[10178]: pbx.c:3794 __ast_pbx_run: Channel
'Console/dsp' sent into invalid extension '555' in context 'usuarios',
but no invalid handler
(!!!)
I'm very sure that the same case (doing the Dial, but Dial is not
working) it goes to n+1.
Sorry for the confusion and thanks for helping.
--
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info Manresa - Barcelona
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Re: [asterisk-users] 3-way calling for IAX channels
On Wednesday 23 July 2008 12:17:26 Steve Davies wrote: 2008/7/23 MFH [EMAIL PROTECTED]: Noah Miller wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf Asterisk supports conferencing without using meetme. In this case you don't have a central dial in number but a single extension can initiate the conference call. Generally this is done the same way as with traditional PSTN service which is that while on a call between two parties, flash the line, dial out to the third party then flash again and all the parties should be connected. I believe that response is slightly misleading - Asterisk does not support conferencing without using meetme, but Zaptel/DAHDI will emulate the PSTN flash/recall facility which looks a bit like a conference. In SIP, IAX, and I believe all other non Zaptel/DAHDI channel types, the endpoint must manage the equivalent of a PSTN flash/recall conference. Anything cross-channel or otherwise more complex does indeed require app_meetme. Given that the OP was referring to IAX, I believe they will need app_meetme. The interesting thing is that Zaptel/DAHDI is using exactly the same conferencing/audio mixing engine as app_meetme. Or more correctly, app_meetme is using the Zaptel/DAHDI engine for audio mixing. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent channel...
I have been looking for the busy-limit directive you mention but cannot find it in any documentation for Asterisk. I can only find something called busy-level which by its description might be what I need. On Wed, 2008-07-16 at 15:20 +, Tariq .. wrote: Try adding busy-limit=1 to your SIP users as it will let the agent to report the Busy as a hint. the call-limit=1 only allows one channel to the agent.. but then if the agent is not busy the queue will try to call them and it will bypass the CW service so the Agent channel will receive the call and drop it immediately. adding the busy-limit=1 will send the busy here hint to the queue when it tries to call it .. and then the queue will try another agent. Salam Tarek Sawah __ Date: Tue, 15 Jul 2008 10:54:34 +1000 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Agent channel... From memory, I have seen something similar done with the SIPPEERS function (curcalls) but it's too fuzzy for me to remember it fully. Paul Hales NTS Carlos Chavez wrote: I have a customer with a small outgoing call center. Usually only 3 to 5 agents online. We are still using Agent/XXX channels in this application on Asterisk 1.4.18. I have an autodialer that is making the outgoing calls and then dropping them into a Queue where all the agents are logged on. My problem is that when an agent makes a call on his/her phone the queue always says that the agent is Not in use. I have call-limit set to 1 on all sip phones that are used for agents but I can see that the queue tries to send a call to the agent. Since the agent has a limit of one the call gets rejected but instead of going back to the queue it is dropped. How can I make sure the agent will show In Use when they make a call? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Use video conversation to talk face-to-face with Windows Live Messenger. Get started. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need help setting up dundi
Hi,
Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below.
tried this on the cli:
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
dundi debug shows this, i have no idea what that means though:
[Jul 24 02:42:39] Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type:
NULL (Command)
[Jul 24 02:42:39] Flags: 00 STrans: 23177 DTrans: 0 [10.10.10.1:4520]
(Final)
[Jul 24 02:42:39] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type:
ACK (Response)
[Jul 24 02:42:39] Flags: 00 STrans: 05678 DTrans: 23177 [10.10.10.1:4520]
(Final)
any mistake on my config?
regards,
ron
asterisk#1 (IP ADDRESS:10.10.10.1)
dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial
[AB:CD:EF:70:E9:DA]
model = symmetric
host = 10.10.10.2
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
sip.conf
[4000]
type=friend
nat=yes
secret=4000
host=dynamic
[priv]
type=peer
context=dundi-priv-canonical
extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})
[dundi-priv-local]
include = dundi-priv-canonical
[dundi-priv-switch]
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
asterisk #2 (IP ADDRESS:10.10.10.2)
dundi.conf
[mappings]
priv = dundi-priv-canonical,0,SIP,[EMAIL PROTECTED],nopartial
[00:1E:8C:AB:CD:EF]
model = symmetric
host = 10.10.10.1
inkey = dundi
outkey = dundi
include = priv
permit = priv
qualify = yes
order = primary
sip.conf
[4001]
type=friend
nat=yes
secret=4001
host=dynamic
[priv]
type=peer
context=dundi-priv-canonical
extensions.conf
[dundi-priv-canonical]
exten = _4XXX,1,Dial(SIP/${EXTEN})
[dundi-priv-local]
include = dundi-priv-canonical
[dundi-priv-switch]
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
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Re: [asterisk-users] Call Recordings...
Would be my guess. J From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eugen Soare Sent: Tuesday, July 22, 2008 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... So basically, He wants all calls recorded, but he wants a sequence that he can push, so that when he rants and raves at a customer, there won't be evidence to say that he did that... :) Just a hunch on that. :) I don't know. Eugen On 7/22/08, Gregory Malsack HYPERLINK mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.4/1566 - Release Date: 7/22/2008 6:00 AM ___ -- Bandwidth and Colocation Provided by HYPERLINK http://www.api-digital.com/; \nhttp://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: HYPERLINK http://www.astricon.net/; \nhttp://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: HYPERLINK http://lists.digium.com/mailman/listinfo/asterisk-users; \nhttp://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble Playing message file via Perl AGI
Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi-exec(background $msg); But of course, this doesn't allow me to capture any keypresses. So I tried this: $agi-stream_file($msg, 0123456789, 0); The console indicates that it's playing the message, but it then skips to the next AGI instruction and nothing gets played. Then I tried to use the get_data() method. It turns out that I had to put two of them in my code, but then the timeout doesn't work and it doesn't capture any keypresses: $result = $agi-get_data($msg, 12, 1); $result = $agi-get_data($msg, 12, 1); Finally, I tried to use the get_option() method that was documented in the module POD file; Perl complains that the method isn't defined: $result = $agi-get_option($msg, 12345, 1); So, what am I missing? I know this works; too many people are doing it. Any ideas? TIA, -- Mike Diehl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recordings...
I'm getting close. The idea is based on the same principal as the link below. Here's what I have done thus far: All calls are recorded via mixmonitor. This is part of the initial dialplan when the call comes in. I then created an application map key sequence that is supposed to run stopmixmonitor. However I am unable to locate examples of syntax on that command. Here is what I have: stoprecording = *8,self/callee,StopMixMonitor, This command syntax does not work and the recording continues on. Can anyone provide direction on this? Thanks, Gregory Malsack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 22, 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Hi Gregory, I found something about recording at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf (second example). If you combine that with a default_recording_enabled (Monitor() call before Dial(), I would expect), that could be used to turn _off_ recording by pressing a key. I would not know though how to protect against the external call party pressing the same key. Best regards Anselm No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Wednesday, July 23, 2008 03:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble Playing message file via Perl AGI
Mike Diehl wrote:
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having trouble getting my program to play a message and wait for a
keystroke.
I am able to use this code to play the file, so I know that the $msg variable
points to a valid sound file:
$result = $agi-exec(background $msg);
But of course, this doesn't allow me to capture any keypresses. So I tried
this:
$agi-stream_file($msg, 0123456789, 0);
The console indicates that it's playing the message, but it then skips to the
next AGI instruction and nothing gets played.
Then I tried to use the get_data() method. It turns out that I had to put
two
of them in my code, but then the timeout doesn't work and it doesn't capture
any keypresses:
$result = $agi-get_data($msg, 12, 1);
$result = $agi-get_data($msg, 12, 1);
Finally, I tried to use the get_option() method that was documented in the
module POD file; Perl complains that the method isn't defined:
$result = $agi-get_option($msg, 12345, 1);
So, what am I missing? I know this works; too many people are doing it. Any
ideas?
TIA,
$agi-get_data is likely what you are looking for. I'm using it
successfully in both standard and FastAGI scripts.
With this sample script:
#!/usr/bin/perl
use Asterisk::AGI;
use strict;
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $digits = $AGI-get_data('tt-monkeys', 1, 1);
$AGI-verbose(We Received $digits,3);
exit;
The CLI outputs:
**
-- Executing [EMAIL PROTECTED]:1] AGI(SIP/1223-090046a8, test.agi)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.agi
-- SIP/1223-090046a8 Playing 'tt-monkeys' (language 'en')
-- test.agi: We Received 4
-- AGI Script test.agi completed, returning 0
When I press 4 when listening to tt-monkeys.
Hope this helps.
- Dave
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Re: [asterisk-users] sometimes extensions can't be called
Hi Sir, Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting ‘qualify=yes’ in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you’ll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be ‘UNREACHABLE’. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25
Re: [asterisk-users] Looking for a more robust Click to Dial/Web Dial solution than AsteriDex (potential for a bounty!)
On Tuesday 22 July 2008 02:58:38 pm Jason Lixfeld wrote: I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between my cell and the other number so I can use my office PBX for company LD, clients see my company's CallerID etc. AsteriDex seems to have almost everything that I'm looking for, but I need something with a few more enhancements and I'm wondering if such a thing exists or if I need this to be custom made. - I need something that can import a phone book from vcards and/or pull names and numbers from an LDAP directory, not just MySQL (I don't even really care about keeping my numbers in AsteriDex's MySQL database). - I need something that, when I hit it with a web browser (specifically, Mobile Safari on my iPhone 3G), will also have a field where I can enter a number manually, incase a number I need to dial isn't in the directory. - I need something that has hooks to customize the CallerID fields. It should have configuration hooks somewhere where I can set a couple of different the CallerID Names and Numbers, then have the option to select which CallerID gets set when the outbound call to the client is made. I have control over the CallerID that gets sent to the Telco. Please advise, and if someone is looking for a few extra bucks, let me know how much you will charge to develop something like this. I can provide a deposit if you are credible. you could look at: http://messinet.com/?page_name=MessinetSecureDirectory it's just my toying around with the concepts, but the basics are there. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadsoft Sip provider
I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I dont need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer? Thanks for any help. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadsoft Sip provider
Quoting Gustavo A Gonzalez [EMAIL PROTECTED]: I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. This is what I use: register = 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368 [broadworks] type=peer host=1.2.3.5 dtmfmode=rfc2833 outboundproxy=1.2.3.4 fromdomain=1.2.3.5 fromuser=3115552368 username=3115552368 authname=3115552368 secret=abcdefghijklmnop canreinvite=no disallow=all allow=gsm allow=g726 allow=ulaw qualify=yes insecure=port,invite context=inbound ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Implementing an Asterisk Server behind a Meridian Norstar
We have an older Meridian Norstar system and are thinking of using Asterisk behind it to use a SIP Voip Provider instead of our local telco. Does anyone make an interface card that can integrate with the digital input of the Meridian. Not the optimal solution, but it allows for the current infrastructure to be retained. Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the magic needed from upgrading from 1.4 to 1.6
On Tue, 2008-07-22 at 13:21 -0400, Jerry Geis wrote: On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote: / // �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite: / Call from 'devcentos5x64_to_ebox4300' to extension 'mediaport_audio_visual' rejected because extension not found. Jerry-- from the console, type dialplan show smvoice-mediaport, and let's verify for certain that it's in there. I'll try to reproduce your problem in my test system here. murf Steve, I get this: dialplan show smvoice-mediaport There is no existence of 'smvoice-mediaport' context Command 'dialplan show smvoice-mediaport' failed. my extensions.conf has a context: ; media [smvoice-mediaport] exten = public_address,1,Goto(smvoice-mediaport-public-address,s,1) #include /etc/asterisk/express.dnis.conf Then express.dnis.conf has: ; This file is generated from MessageNet EMACS ; Phone Caller ID DNIS Manager screen ; MMAUDIO : EBOX 4300 - exten = mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1) [smvoice-mediaport-audio-visual] exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Hangup Not seeing what the problem is here. especially since 1.2 and 1.4 both work. Jerry Jerry-- I've opened a bug in your behalf at http://bugs.digium.com/view.php?id=13144 Please follow the above link and hit the 'monitor issue' button there, and it will send you an email whenever the issue has updates. I don't know if you created an account on bugs.digium.com, but if you have not, it would be a good idea (and time) to register. I've been pounding my head against the wall with a subtle bug that I *think* I've fixed; I've decided to commit the fix and close the above bug, but I realize full well that it may not be a fix to your problem! So, here is the plan: if after I close 13144, and you update your trunk/1.6 version of asterisk, and you still have the problem, then re-open 13144, and further discussion on this problem will occur via this bug report. The bug I fixed involved a memory leak in the dialplan structures, which has resulted, for me, in: 1. missing contexts 2. crashes on loading 3. crashes during 'stop gracefully' I found the problem on a code review, and valgrind verified that in some circumstances, it was happening. Fixing it cleared up all the weird affects. But then again, I managed to intensify the bug by having lotsa code in both extensions.conf, and in extensions.ael, and having the smvoice-mediaport-audio-visual context in BOTH files. The inclusion did not affect the results; if I included express.dnis.conf, or just pasted its contents in place of the '#include..', it didn't matter. So, please monitor that bug, and let me know if all is well. murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
Does anyone make an interface card that can integrate with the digital input of the Meridian. Not the optimal solution, but it allows for the current infrastructure to be retained. By digital input do you mean a T1 interface? If so then yes several T1 interfaces are available. However I think you mean is there a gateway to use the Meridian/Norstar phones with Asterisk. If so, yes there is a company that makes a gateway to use the Nortel p-phones with a SIP based system. However past experience has shown that for the less than the cost of the gateway, I could replace the phones with IP phones and eliminate another point of failure and the hassle of configuring it. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
By digital input do you mean a T1 interface? If so then yes several T1 interfaces are available. However I think you mean is there a gateway to use the Meridian/Norstar phones with Asterisk. If so, yes there is a company that makes a gateway to use the Nortel p-phones with a SIP based system. However past experience has shown that for the less than the cost of the gateway, I could replace the phones with IP phones and eliminate another point of failure and the hassle of configuring it. John, Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the sounds of it, its not worth it. I guess the only thing I would need is a small switch in each office then as we only have one run of cat-5e to each office. Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
Joseph L. Casale wrote: Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the sounds of it, its not worth it. I guess the only thing I would need is a small switch in each office then as we only have one run of cat-5e to each office. The company I think John was referring to is Citel, and they do make gateways that translate between legacy digital PBXs and SIP. But I would tend to agree that the cost isn't worth it, nor does it provide a permanent solution. How big is this installation? Depending on the number of seats, you could probably get additional, upgraded cabling done to each office and buy handsets, and still make off with less cost than you would trying to adapt the Meridian--quite imperfectly, at that. Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? Not as far as I know, since a desktop application PC is the highest-bandwidth device manufacturers assume anyone would want to uplink through a phone. Although, oddly enough, a lot of them can do VLAN trunking, etc. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect Asterisk PBX to Traditional PBX and retain functionality
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It's possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I Disable call-waiting
Rob Hillis wrote: If there were another usable softphone not tied to a specific platform (such as Cisco CallManager) that had proper support for Plantronics CS60 USB headsets, I would have made the switch ages ago. Does the eyeBeam have a textual local configuration file? If so, you could probably roll your own autoprovisioning with a script that grabs a config for a specific client IP address/MAC address/et., perhaps even packaging it into a custom installer. And build a wrapper around the executable that fetches the config every time the phone starts. I'm not the person to ask on how to do this, as a Linux guy, but I am sure it is possible with a little cajoling, although of course I would opine that a UNIX-style runtime environment would lend itself to that sort of tomfoolery much easier. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble Playing message file via Perl AGI
AGI can wrap calls to any dial plan applications; have you tried calling Background() and Read() that way? Mike Diehl wrote: Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi-exec(background $msg); But of course, this doesn't allow me to capture any keypresses. So I tried this: $agi-stream_file($msg, 0123456789, 0); The console indicates that it's playing the message, but it then skips to the next AGI instruction and nothing gets played. Then I tried to use the get_data() method. It turns out that I had to put two of them in my code, but then the timeout doesn't work and it doesn't capture any keypresses: $result = $agi-get_data($msg, 12, 1); $result = $agi-get_data($msg, 12, 1); Finally, I tried to use the get_option() method that was documented in the module POD file; Perl complains that the method isn't defined: $result = $agi-get_option($msg, 12345, 1); So, what am I missing? I know this works; too many people are doing it. Any ideas? TIA, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Hi Sir Thanks for your reply, since i don't know how to setup DUNDi, what i did for now is create a sip peer between the 2 servers and just use the regserver on the realtime db. but now with that setup i cant play the music on hold of the extension i'm calling to, e.g i'm 118102 i called 118103 1182102 has moh class moh-118102 and 118103 has class moh-118103. if the call is on the same server i have no issues moh plays the class of the user, but when the extension is on the other server and i put it on hold, it always plays the class default, anyway i will try to figure that one out also, thanks again to all your reply. regards, nhadie Noah Miller wrote: Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
Alex Balashov wrote: Although, oddly enough, a lot of them can do VLAN trunking, etc. Not odd at all as far as I'm concerned - I know a number of places that segregate LAN traffic from VoIP traffic using multiple VLANs over the one physical link. VLANs would be the best solution (short of running multiples cables for PC and phone) to achieve this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing an Asterisk Server behind a MeridianNorstar
Not odd at all as far as I'm concerned - I know a number of places that segregate LAN traffic from VoIP traffic using multiple VLANs over the one physical link. VLANs would be the best solution (short of running multiples cables for PC and phone) to achieve this. I would have about 30 phones I think over 6-12 lines. Vlans would be a must as I would surely be using the same network infrastructure. I will keep hunting... A small switch in each office might not be a big deal. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recordings...
I resolved this problem. The key was to get the right combination of self/callee and peer/caller. Read the instructions regarding the application map very closely. My problem was that I was not running the StopMixMonitor command against the proper channel. Even though mixmonitor records both channels simultaneously, the recording is only assigned to 1 channel and you have to run the command against the originating channel of the call. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack Sent: Wednesday, July 23, 2008 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... I'm getting close. The idea is based on the same principal as the link below. Here's what I have done thus far: All calls are recorded via mixmonitor. This is part of the initial dialplan when the call comes in. I then created an application map key sequence that is supposed to run stopmixmonitor. However I am unable to locate examples of syntax on that command. Here is what I have: stoprecording = *8,self/callee,StopMixMonitor, This command syntax does not work and the recording continues on. Can anyone provide direction on this? Thanks, Gregory Malsack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 22, 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recordings... Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack: Hello, My boss is asking me to setup the asterisk server to record all calls. (Simple). However, he wants to be able to enter a key sequence during the call to stop the recording. Any ideas on how I would do that? Hi Gregory, I found something about recording at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf (second example). If you combine that with a default_recording_enabled (Monitor() call before Dial(), I would expect), that could be used to turn _off_ recording by pressing a key. I would not know though how to protect against the external call party pressing the same key. Best regards Anselm No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.5.5/1568 - Release Date: 7/23/2008 6:55 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Asterisk PBX to Traditional PBX and retain functionality
Ricardo Melendez wrote: Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It’s possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez I have used the FLASH command in Asterisk to generate flashes...is that what you are asking? later, PaulH Asterisk says: *CLI show application Flash -= Info about application 'Flash' =- [Synopsis] Flashes a Zap Trunk [Description] Performs a flash on a zap trunk. This can be used to access features provided on an incoming analogue circuit such as conference and call waiting. Use with SendDTMF() to perform external transfers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on Codecs
On Thu, 17 Jul 2008, Nhadie wrote: Hi, I'm testing using the free g723 codecs and i have successfully installed them. g723 g723 - gsm 9 ulaw 9 alaw 9 g726 9 adpcm 9 slin 8 lpc1010 g72910 speex - ilbc10 i also set my pap2's to use G723. I'm sending the call to an AS5400 with multiple E1's on it. problem i noticed is that my CPU went up to 65% usage. From 4% when i was using ulaw. does asterisk still do codec translation even if the g723 codec is installed on it? this is the result from top command: %CPU %MEM 16829 asterisk 15 0 42500 22m 5520 S 65 2.6 91:21.93 asterisk i have 55 simultaneous users testing it. as far as my understanding, when g723 is installed on the asterisk there should be no codec translation done on the server thus not utilizing that much CPU. am i understanding it correctly? thanks in advanced AFAIK, when asterisk switches calls from one format(i.e g723 etc) to other format(PCM), it needs to transcode the streams and that's what the codec that you installed are for. if both sides of the calls use same codec format, you wouldn't need any codec as asterisk would simply pass thru the streams. And I'm sure E1's won't use g723. Manoj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tomato = One Way Audio; Linksys = OK ????
Hey Guys, New TrixBox. For some reason it works just fine behind a WRT54GL with latest version of stock Linksys firmware. However, when using a GL with Tomato firmware, can't hear the ringing or audio from the called party. Yes, ports 10,000 - 20,000 and 5004 - 5082 are open. Yes, these lines are in sip.conf : ;;; externip=20x.15x.18x.1xx localnet=192.168.1.0/255.255.255.0 #include sip_nat.conf ;;; Yes, this line is in sip_nat.conf: ;;; nat=yes ;;; Yes, these lines are in rtp.conf: ;;; [general] rtpstart=1 rtpend=2 ;;; Any other tips or ideas? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble Playing message file via Perl AGI
David,
What you sent me is almost exactly what I had, which indicated that that part
of my code was correct. So, I moved that block of code to the top of my
program and it worked. Eventually, I found a debug print() statement that I
had forgotten to take out. Once it was gone, my code worked as expected.
Thank you for your time.
Mike.
On Wednesday 23 July 2008 01:48:14 pm David Van Ginneken wrote:
Mike Diehl wrote:
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having trouble getting my program to play a message and wait for
a keystroke.
I am able to use this code to play the file, so I know that the $msg
variable points to a valid sound file:
$result = $agi-exec(background $msg);
But of course, this doesn't allow me to capture any keypresses. So I
tried this:
$agi-stream_file($msg, 0123456789, 0);
The console indicates that it's playing the message, but it then skips to
the next AGI instruction and nothing gets played.
Then I tried to use the get_data() method. It turns out that I had to
put two of them in my code, but then the timeout doesn't work and it
doesn't capture any keypresses:
$result = $agi-get_data($msg, 12, 1);
$result = $agi-get_data($msg, 12, 1);
Finally, I tried to use the get_option() method that was documented in
the module POD file; Perl complains that the method isn't defined:
$result = $agi-get_option($msg, 12345, 1);
So, what am I missing? I know this works; too many people are doing it.
Any ideas?
TIA,
$agi-get_data is likely what you are looking for. I'm using it
successfully in both standard and FastAGI scripts.
With this sample script:
#!/usr/bin/perl
use Asterisk::AGI;
use strict;
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $digits = $AGI-get_data('tt-monkeys', 1, 1);
$AGI-verbose(We Received $digits,3);
exit;
The CLI outputs:
**
-- Executing [EMAIL PROTECTED]:1] AGI(SIP/1223-090046a8, test.agi)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.agi
-- SIP/1223-090046a8 Playing 'tt-monkeys' (language 'en')
-- test.agi: We Received 4
-- AGI Script test.agi completed, returning 0
When I press 4 when listening to tt-monkeys.
Hope this helps.
- Dave
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--
Mike Diehl
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Re: [asterisk-users] Implementing an Asterisk Server behinda MeridianNorstar
Well, I am not sure what is needed to interface between the two. I hoped there was something you could use and from the Joseph, Now I'm pretty sure we are not talking about the same things. Let me see if I have the correct picture in my head. I now think you have a Norstar in one office and an asterisk system in another office and want to allow them to send calls between them. Is this correct? Do they make phones with a gig switch in them? I am told there are phones with 100meg switches in them? The new Polycom 670 has a gig interface but at this point I'm not sure why you need that. Are you thinking that if the Norstar phones and lines can't be used, that you would need the phone to have a switch to share the Ethernet connection? Sorry for the confusion but I just want to make sure I know what you need before making a recommendation. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
