Re: [asterisk-users] OT - DECT SIP Phones
- "--[ UxBoD ]--" wrote: | - "Alan Lord (News)" wrote: | | | On 17/10/09 15:02, --[ UxBoD ]-- wrote: | | > Hi, | | > | | > I have three Snom M3s at the moment but getting pretty fed up with | | the issues :( I am UK based and would be interested to hear of | other | | peoples recommendations. Key features :- | | > | | > * VM Notification | | > * Good Range | | > * G729 codec support | | > * Common/Private Address Books per Handset(s) | | | | Siemens Gigaset: | | | http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ | | | | One of the most popular posts on my blog over the last 1 1/2 years. | It | | | | still gets lots of hits from people looking for info on them. | | | | FYI We have two sets in our network - they haven't missed a beat | since | | | | installation. | | | | HTH | | | | Alan | | | | | | S685 set turned up the other day and have had a good chance to try it out ... Voice quality is definitely superior to the M3 though I guess that will be addressed in the M9 with G722 support. Impressions of the Siemens phone :- Pros Great Voice Quality Easy to configure Good range Ability to transfer address books Good build quality Cons Keys feel fiddly Keys response time when dialling is very slow When call connection is made a audible beep is heard which cuts of the first .5 second of a VM message DTMF does not all transfer correctly On the whole it seems a very capable phone and very well laid out. For my personal needs it just does not feel right with how the keys react. Hopefully the M9 will be released very shortly so I can make a comparison. Thanks to everyone's replies. Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT SIP Phones
On 23/10/09 08:34, --[ UxBoD ]-- wrote: > > S685 set turned up the other day and have had a good chance to try it out ... > Voice quality is definitely superior to the M3 though I guess that will be > addressed in the M9 with G722 support. > > Impressions of the Siemens phone :- > > Pros > > Great Voice Quality > Easy to configure > Good range > Ability to transfer address books > Good build quality > > Cons > > Keys feel fiddly > Keys response time when dialling is very slow > When call connection is made a audible beep is heard which cuts of the first > .5 second of a VM message > DTMF does not all transfer correctly > > On the whole it seems a very capable phone and very well laid out. For my > personal needs it just does not feel right with how the keys react. I agree with the response time on the keys, it does seem to be quite slow but I have got used to that now - after 1 1/2 years :-) Will the M9 also have an analogue interface? That is one of the main reasons for my choosing this phone. It's a great dual home/home-office phone because of that. Cheers Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriCon videos: a question of method
As others have said, John, Viddler is good. If you have any shorter-than 10 minute videos, you might put them on YouTube as well for the sheer exposure and then add something pointing to a Viddler URL for "additional, longer content". /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriCon videos: a question of method (Robin)
Hi Michael, cool that you like viddler.com :). Currently downloading your uploads to watch at home from my ps3 (convenience of the couch). Cheers, Robin On Fri, Oct 23, 2009 at 08:33, Michael Collins wrote: > Robin, > > Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon > videos to it right now. > > John, so far I'd have to give viddler.com two thumbs up. I'm adding my > stuff here: > http://www.viddler.com/explore/cluecon > > Your ClueCon presentation should show up some time on Friday. I've noticed > that there's a little bit of a lag time between upload and video being > available for viewing but that's completely reasonable under the > circumstances. Let us know what you decide. > > -MC > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT SIP Phones
- "Alan Lord (News)" wrote: | On 23/10/09 08:34, --[ UxBoD ]-- wrote: | | > | > S685 set turned up the other day and have had a good chance to try | it out ... Voice quality is definitely superior to the M3 though I | guess that will be addressed in the M9 with G722 support. | > | > Impressions of the Siemens phone :- | > | > Pros | > | > Great Voice Quality | > Easy to configure | > Good range | > Ability to transfer address books | > Good build quality | > | > Cons | > | > Keys feel fiddly | > Keys response time when dialling is very slow | > When call connection is made a audible beep is heard which cuts of | the first .5 second of a VM message | > DTMF does not all transfer correctly | > | > On the whole it seems a very capable phone and very well laid out. | For my personal needs it just does not feel right with how the keys | react. | | I agree with the response time on the keys, it does seem to be quite | slow but I have got used to that now - after 1 1/2 years :-) | | Will the M9 also have an analogue interface? That is one of the main | reasons for my choosing this phone. It's a great dual home/home-office | | phone because of that. | | Cheers | | Al Hi Alan, The feature set of the M9 does not appear to be available anywhere yet, unfortunately as it is being released late this month, so not sure about the analogue interface. Personally I have a TDM card in my Asterisk server so that feature in the phone is not on my list of requirements. I type pretty quick on a handset so the slow response does hinder me a bit which is why it went on my cons list. The Siemens is a good phone as I expect, hopefully ;), the M9 will be. As always I guess it will come down to personal preference. I just hope when the M9 is released it does not have all the issues the M3 had when it was first launched. Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] carefulwrite: write() returned error:Brokenpipe
Sorry it was the AGI STATUS variable, that I forgot to return. Best regards, Josip -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, October 22, 2009 11:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] carefulwrite: write() returned error:Brokenpipe On Thu, 22 Oct 2009, Tzafrir Cohen wrote: > On Thu, Oct 22, 2009 at 01:30:31PM -0700, Steve Edwards wrote: >> On Thu, 22 Oct 2009, Danny Nicholas wrote: >> >>> Sorry about the "top post" (OUTLOOK) - >> >>> Thanks for the framework. It's easier to learn from a starting point >>> than scratch. I'm not crazy about writing 1000 lines of C to do 30 >>> lines of PERL, but if it makes my system fly, so be it. >> >> If you discard my comments and account for my "open and explicit" coding >> style, it's less than 200, but yes, Perl is a bit more "dense." >> >> I don't have any experience with them, but there are 2 C AGI libraries >> available -- cagi and quivr. > > There's also Asterisk::AGI in CPAN. Except, the OP expressed interest in writing AGIs in C. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
Steve Edwards writes: > atftpd can do PCRE substitutions to transform a requested file name into > something else. I've not used this facility, but I'm guessing you could > transform: > > SIPDefault.cnf -> cisco/SIPDefault.cnf > sip.cfg -> polycom/sip.cfg > spa841.cfg -> sipura/spa841.cfg Cute, but all that accomplishes is renaming. I want to run a script which returns a different configuration based on the file name (and possibly the client IP address). Unfortunately there is also no UserAgent-header in TFTP... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC with DAHDI and Linksys/Sipura
On Thu, Oct 22, 2009 at 06:14:29PM -0600, Joseph wrote: > On 10/23/09 01:40, Tzafrir Cohen wrote: > >So all you have to do is implement Asterisk (or whatever) on that > >Linksys device, provide DAHDI drivers for its FXS and FXO, and you're > >done. Should be easy. > > Well, I've never used zaptel for Linsys/Sipura adapter as zaptel does not > make any difference on their operations. > and I'm not sure that DAHDI drivers exist for Linksys/Sipura adapters. > DAHDI stands for "Digium Asterisk Hardware Device Interface" so I don't > see how can I take advantage of DAHDI with Sipura adapters. Which is only because nobody wrote it yet. (And no, I was not seriously suggesting that you do that) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crashes when calling gtalk user
Hi all,
I'm using Asterisk 1.4.26.2. Every time I call a gtalk user, Asterisk
crashes:
-- Executing [6...@inbound:1] NoOp("SIP/8-084894d8", "jabbertest")
in new stack
-- Executing [6...@inbound:2] Dial("SIP/8-084894d8",
"Gtalk/asterisk/pippopi...@gmail.com") in new stack
Segmentation fault
Is anybody experiencing the same? Found any workaround?
Thank you.
Giorgio Incantalupo
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[asterisk-users] How to generate 183 Session Progress
Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer The one that doen't work: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces -- -- -- Marc LEURENT lf...@leurent.eu ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording management for IVR
Hello everyone. I have a client with specific requirement, here's the scenario: Call comes in Ivr menu, press 1 for new record 2 for existing 3 for operation blabla.. on pressing 1, list of 5 categories A,B,C,D .. when customer selects a category a 4 digit pin needs to be generated and a recording be recorded in a specific folder A,B,C,D or E. then this 4 digit number is mailed ( can be done easily ). On pressing 2 at main ivr, read 4 digits and play back the corresponding file from given 5 directories. What is the easiest way to get this done ? Php agi or just dialplan hacks ? How can i geenrate a 4 digit unique pin each time for the file ? TIA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording management for IVR
hello AGI is a good option to handle such complexities On Fri, Oct 23, 2009 at 6:33 PM, Mail list wrote: > Hello everyone. I have a client with specific requirement, here's the > scenario: > > Call comes in > Ivr menu, press 1 for new record 2 for existing 3 for operation blabla.. > on pressing 1, list of 5 categories A,B,C,D .. when customer selects a > category a 4 digit pin needs to be generated and a recording be recorded in > a specific folder A,B,C,D or E. > then this 4 digit number is mailed ( can be done easily ). > > > On pressing 2 at main ivr, read 4 digits and play back the corresponding > file from given 5 directories. > > > What is the easiest way to get this done ? Php agi or just dialplan hacks ? > How can i geenrate a 4 digit unique pin each time for the file ? > > > TIA. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
> Steve Edwards writes: > >> atftpd can do PCRE substitutions to transform a requested file name >> into something else. I've not used this facility, but I'm guessing you >> could transform: >> >> SIPDefault.cnf -> cisco/SIPDefault.cnf >> sip.cfg -> polycom/sip.cfg >> spa841.cfg -> sipura/spa841.cfg > On Fri, 23 Oct 2009, Benny Amorsen wrote: > Cute, but all that accomplishes is renaming. I want to run a script > which returns a different configuration based on the file name (and > possibly the client IP address). Unfortunately there is also no > UserAgent-header in TFTP... So, break out your H&S or your wallet :) How about adding an option to atftpd to enable "check the execute bit" on the requested file. If the file has the execute bit, set the MAC, IP Address, etc. as environment variables, exec the file, and return the output to the requester? Then, you could write (for example) spa841.cfg as a shell script which have access to the MAC, etc. and could run sed/replace/preprocessor/m4 and create the "file" on the fly. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interfacing asterisk with a legacy PBX
PATRICK KANGETHE wrote: > I want to interface asterisk with a legacy pbx that has around 23 > extensions through my 8 fxs card, how do i work around this? > Hint: I have already terminated 8 extensions from the legacy PBX, i > was thinking whether i can peer the extensions from the PBX i.e like 5 > extensions be peered to one extension connecting to the fxs? How can i > do this? > > Thanks in advance, > > Are you planning to get rid of the legacy PBX completely? Or is Asterisk going to be a second PBX? I am going to assume you are replacing the legacy PBX. You can setup analog extensions so that you have multiple phones on each FXS channel. But they will be like a party line. If you put 6 phones on one FXS, all 6 ring at the same time, only one person can use that extension at a time. However you can add SIP phones to Asterisk and each can have their own extension instead. It just requires cat 5 cable back to a switch for each phone. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to generate 183 Session Progress
If the outgoing channel receives progress indication from the far end (e.g. ISDN PROGRESS message or 183 response from an ITSP) then Asterisk will relay the progress message. If there is no progress indication received - that means that early media is not available - Asterisk does not send 183 as this would make the client listing for early media although it is not available. regards klaus Marc Leurent schrieb: > Hello everybody, > I have 2 users connected on the same Asterisk server that are connected with > 2 different Asterisk servers. > > For outgoing calls, one in receiving 183 Session Progress and the other not! > Do you have any idea why? > Thanks. > > I have tried to understand by myself and in their INVITE they have almost the > same Allow and Supported SIP Headers > > The one that works: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > > The one that doen't work: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb: > When Asterisk establish a call through an outbound trunk, Is there any > way I can know who hang up the call first? The caller or the party called? you could use the 'g' option of the Dial command together with some logic in the hangup extensions regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
We have queuemetrics and it does that
Here is some of the logic - (Obviously this wont work for you right out
of the box but you should be able to decipher the logic...)
[qm-queuedial]
; We use a global variable to pass values back from the answer-detect
macro.
; STATUS = U unanswered
;= A answered(plus CAUSECOMPLETE=C when callee hung up)
; The 'g' dial parameter must be used in order to track callee
disconnecting.
; Note that we'll be using the 'h' hook in any case to do the logging
when channels go down.
; We set the CDR(accountcode) for live monitoring by QM.
;
exten => s,1,NoOp,Outbound call -> A:${QDIALER_AGENT}
N:${QDIALER_NUMBER} Q:${QDIALER_QUEUE} Ch:${QDIALER_CHANNEL}
exten => s,n,Set(CDR(accountcode)=QDIALAGI)
exten => s,n,Set(ST=${EPOCH})
exten => s,n,Set(GM=QDV-${QDIALER_AGENT})
exten => s,n,Set(GLOBAL(${GM})=U)
exten => s,n,Set(GLOBAL(${GM}ans)=0)
exten =>
s,n,Macro(queuelog,${ST},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT},C
ALLOUTBOUND,-,${QDIALER_NUMBER})
exten =>
s,n,Dial(${QDIALER_CHANNEL},300,gM(queuedial-answer^${UNIQUEID}^${GM}^${
QDIALER_QUEUE}^${QDIALER_AGENT}^${ST}))
exten => s,n,Set(CAUSECOMPLETE=${IF($["${DIALSTATUS}" = "ANSWER"]?C)})
; Trapping call termination here
exten => h,1,NoOp( "Call exiting: status ${GLOBAL(${GM})} answered at:
${GLOBAL(${GM}ans)} DS: ${DIALSTATUS}" )
exten => h,n,Goto(case-${GLOBAL(${GM})})
exten => h,n,Hangup()
; Call unanswered
exten => h,n(case-U),Set(WT=$[${EPOCH} - ${ST}])
exten =>
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},ABANDON,1,1,${WT})
exten => h,n,Hangup()
; call answered: agent/callee hung
exten => h,n(case-A)i,Set(COMPLETE=${IF($["${CAUSECOMPLETE}" =
"C"]?COMPLETECALLER:COMPLETEAGENT)})
exten => h,n,Set(WT=$[${GLOBAL(${GM}ans)} - ${ST}])
exten => h,n,Set(CT=$[${EPOCH} - ${GLOBAL(${GM}ans)}])
exten =>
h,n,Macro(queuelog,${EPOCH},${UNIQUEID},${QDIALER_QUEUE},${QDIALER_AGENT
},${COMPLETE},${WT},${CT})
exten => h,n,Hangup()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus
Darilion
Sent: Friday, October 23, 2009 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb:
> When Asterisk establish a call through an outbound trunk, Is there any
> way I can know who hang up the call first? The caller or the party
called?
you could use the 'g' option of the Dial command together with some
logic in the hangup extensions
regards
klaus
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[asterisk-users] GUI for asterix management
Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7921
How can i please register Cisco 7921 (Skinny) wireless phone on Asterisk. Thanks _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7921
On 17:25, Fri 23 Oct 09, Torintino T wrote: > > > How can i please register Cisco 7921 (Skinny) wireless phone on Asterisk. You will have to configure chan_skinny using /etc/asterisk/skinny.conf Have a look at the sample file, it has plenty of docs in it. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer aficionados are both called users?" ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange IAX2 / Iaxmodem problem
Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE! Time: 3 [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now REACHABLE! Time: 3 The strangeness comes when I debug the iax2 channel (iax2 set debug): the events(missed PONG I guess) that lead to the UNREACHABLE state don't appear on the console and on the log I get only 10% of the total state change messages I was getting first. Disabling the debug, the lines reappear as before. The state is changed from UNREACHABLE to REACHABLE after 10 seconds(given by qualifyfreqnotok), as documented above. Here is what I'm using to do my tests: Debian 5.0.2 Kernel 2.6.26 Asterisk 1.4.21 (tried 1.6 but the problem it's still there) Iaxmodem-1.1.1 DAHDI 2.2.0 Sangoma A104 CARD and to be safe I loaded the dahdi_dummy too *CLI> dahdi show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 wanpipe2 card 1 RED0 0 0 wanpipe3 card 2 RED0 0 0 wanpipe4 card 3 RED0 0 0 DAHDI_DUMMY/1 (source: HRtimer) 1UNCONFIGUR 0 0 0 The unreachable part is rather odd given that the two apps are on the same host. Reading some articles on the subject I found that this behavior can become a problem if the call arrives when the iaxmodem peer is in the unreachable state. Is there anyone that has an idea on what might be causing this? Sorry for the English and thanks in advance for any help you can provide, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange IAX2 / Iaxmodem problem
Alexandru Oniciuc wrote: > > Hello. > > I’m having a strange problem with the IAX2 channel and IAXmodem and I > was hoping to get some light from someone in these lists. > > On my logs and on the console I’m getting a bunch of lines with: > > [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now > UNREACHABLE! Time: 3 > > [Oct 23 14:26:28] NOTICE[4413] chan_iax2.c: Peer 'XXX' is now > REACHABLE! Time: 3 > Set qualify=no in your iax modem conf (iax.conf) Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriCon videos: a question of method
Something that didn't require flash (works on the iPhone) would be nice. blip.tv may be an option. On Oct 22, 2009, at 3:34 PM, John Todd wrote: > > I'm doing some quick research on how to get our videos from AstriCon > available in a "reasonable" format that allows easy viewing, reduces > our bandwidth costs, and allows good tracking for who/where/what is > viewing the videos. > > YouTube seems to have a very nice set of tools and statistics > collection methods, and might be perfect EXCEPT Their main > limitation right now seems to be that they limit videos to 10 minutes, > which clearly doesn't work for our longer presentations. I could > "patch" them together in multiple 10-minute sessions, but... ugh. > UGH. > > There are other video sites out there - lots, actually. I could spend > hours digging through them all, or hopefully ask here on the list and > have some people give me prior experiences based on their expertise > with hosted video solutions. > > Requirements (not exhaustive list): > - free or very close to free (we'll pay, but not a lot) > - good statistical collection (who is linking? how many views? how > much video watched each view? where do people stop?) > - reasonably easy interaction (good upload tools, good UI) > - good viewing experience from North America, Europe, Asia > > Before anyone suggests it, I'm not interested in Torrent-based > distribution for various political reasons. I've started to look at > Flowplayer, which is appealing due to it's OSS nature and > customization capability, but it leaves us holding the bandwidth bill > (which may not be horrible, but it's a concern.) > > What are your experiences? I can't say we'll end up actually using > what you might think is best, but I'm very interested to hear what > everyone might suggest for distributing Asterisk-focused video > material. > > In the interests of keeping this thread from getting out of control, > please limit yourself to factual, content-rich posts. "I hate > YouTube" or "Why didn't you film blah" is something we can discuss > off- > list. > > JT > > --- > John Todd email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriCon videos: a question of method (Robin)
I was at the site and could not download the videos -- could that be enabled? Robin wrote: > Hi Michael, > > cool that you like viddler.com :). Currently downloading your uploads to > watch at home from my ps3 (convenience of the couch). > > Cheers, > > Robin > > On Fri, Oct 23, 2009 at 08:33, Michael Collins wrote: > > > Robin, > > > > Thanks for the viddler.com suggestion! I'm uploading all of the ClueCon > > videos to it right now. > > > > John, so far I'd have to give viddler.com two thumbs up. I'm adding my > > stuff here: > > http://www.viddler.com/explore/cluecon > > > > Your ClueCon presentation should show up some time on Friday. I've noticed > > that there's a little bit of a lag time between upload and video being > > available for viewing but that's completely reasonable under the > > circumstances. Let us know what you decide. > > > > -MC > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > Alternatives: > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP to Cisco IAD2430 Series?
Hi All, I tried doing SIP from Asterisk to the old Cisco 2400 series IAD's but could not get signalling reliable or DTMF either. I do SIP to Cisco routers with PRI cards in quite a bit, but I guess the old IAD SIP stack is not a robust at the router sip stack so I just could not get it working. I'm wondering if anyone has tried the new Cisco 2430 series IAD's and have been successful and reliable, care to share your experience and sample configs? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hangup from which side
if you are debugging visually then look at SIP BYE message ... who sent it first and on PRI who sent the DISCONNECT message first. if you need to know that in the dialplan ... then if the originating channel hanged up then the dialplan should stop executing and go straight to h,1 even if Dial(,,g) is used also there is a channel variable HANGUPCAUSE and you can check what it does on the next step with Dial(,,g) and on h,1 ... since I don't know :) Martin On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH wrote: > When Asterisk establish a call through an outbound trunk, Is there any way I > can know who hang up the call first? The caller or the party called? > > > > Thanks. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to generate 183 Session Progress
You can call application Progress() from within dialplan and it will cause the Asterisk to send a SIP reply 183 on the call that came to Asterisk. Martin On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent wrote: > Hello everybody, > I have 2 users connected on the same Asterisk server that are connected with > 2 different Asterisk servers. > > For outgoing calls, one in receiving 183 Session Progress and the other not! > Do you have any idea why? > Thanks. > > I have tried to understand by myself and in their INVITE they have almost the > same Allow and Supported SIP Headers > > The one that works: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > Supported: replaces, timer > > The one that doen't work: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > > -- > -- -- > Marc LEURENT > lf...@leurent.eu > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC with DAHDI and Linksys/Sipura
On Fri, Oct 23, 2009 at 5:41 AM, Tzafrir Cohen wrote: >> Well, I've never used zaptel for Linsys/Sipura adapter as zaptel does not >> make any difference on their operations. >> and I'm not sure that DAHDI drivers exist for Linksys/Sipura adapters. >> DAHDI stands for "Digium Asterisk Hardware Device Interface" so I don't >> see how can I take advantage of DAHDI with Sipura adapters. > > Which is only because nobody wrote it yet. writing a DAHDI driver for a SIP device would be counter productive ... you'd have to implement a big chunk of the SIP stack in the kernel ... it'd be easier to write a virtual dahdi driver for one channel that will loop back to itself on the opposite end... then you connect opposite end to B leg via another Dial so again ... SIP device -- incoming call --> Asterisk -- Dial --> virtual dahdi channel1 -- virtual loopback --> virtual dahdi channel2 -- (incoming call) --> Asterisk -- Dial --> SIP destination or whatever that way EC can work both ways or you can turn it on one way only ... via some dialplan application Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crash with app_mixmonitor
Hello All,
I posted a bug on the 14th of this month, and haven't heard anything
back. However, I've since discovered that the problem is not in
chan_iax.c as I originally thought, it's actually app_mixmonitor.c.
Basically when I use 1.4.26.2 with an ilbc codec between two asterisk
servers trunked via IAX, with mixmonitor Asterisk crashes on me.
Here's a link to the post:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=16070.
Can someone possibly assign it to the right application.
I started looking at 1.4.24 which didn't crash, and upped my revisions
until I found the problem. The bug was introduced in revision 204012
of 1.4, here's the info from the changelog:
2009-06-29 15:04 + [r204012] Mark Michelson
* apps/app_mixmonitor.c: Place unlock of mutex in an else block so
that it does not get unlocked twice. (closes issue 0015400)
Reported by: aragon
Here is a diff on the two app_mixmonitor.c files:
--- ./asterisk-204000/apps/app_mixmonitor.c 2009-10-23 13:40:21.0 -0400
+++ ./asterisk-204012/apps/app_mixmonitor.c 2009-10-23 14:03:27.0 -0400
@@ -35,7 +35,7 @@
#include "asterisk.h"
-ASTERISK_FILE_VERSION(__FILE__, "$Revision: 201423 $")
+ASTERISK_FILE_VERSION(__FILE__, "$Revision: 204012 $")
#include
#include
@@ -273,8 +273,9 @@
ast_writestream(*fs, cur);
}
}
+ } else {
+ ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
}
- ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* All done! free it. */
ast_frame_free(fr, 0);
Any chance someone can look at this? I've noticed it happens with
1.6.0.15 as well. I'm going to see if I can find out where it's
introduced in 1.6 as well. Thanks.
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[asterisk-users] SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all codecs allow=siren14; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/9. a=fmtp:109 profile-level-id=42800d; max-mbps=4; max-fs=1792; max-br=1025. a=rtpmap:34 H263/9. a=fmtp:34 CIF4=1;CIF=1; Thanks in advance for any tips, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to announce the agent answering in a queue to the caller
Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Polycom has a softphone? Is it any good? I've never seen it on their site before. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning Sent: Friday, October 23, 2009 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIREN14 call setup and record/playback I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all codecs allow=siren14; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/9. a=fmtp:109 profile-level-id=42800d; max-mbps=4; max-fs=1792; max-br=1025. a=rtpmap:34 H263/9. a=fmtp:34 CIF4=1;CIF=1; Thanks in advance for any tips, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Tom Browning wrote: > > I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of > Asterisk and I'm trying to get it to accept a SIREN14 call from > Polycom's softphone. Having trouble with SDP negotiation, I want to > only allow SIREN14 and nothing else. I also want to record and playback > files, any tips on what the Record function parameters should be? First, don't enable any codecs labeled 'SIREN7' or 'SIREN14' on the Polycom phone; those are pre-standard names, and they work slightly differently than the ITU standardized codecs. > In sip.conf I have: > > disallow=all ; First disallow all codecs > allow=siren14; Is this the right name? Yes, this is correct. Asterisk would also accept 'g.7221c', which is the ITU standardized name. > And the INVITE comes from the Polycom softphone with an SDP of: > > ... > User-Agent: Polycom VV 8.0.4.4035. > ... > m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. > a=rtpmap:99 SIREN14/16000. > a=fmtp:99 bitrate=48000. > a=rtpmap:98 SIREN14/16000. > a=fmtp:98 bitrate=32000. > a=rtpmap:97 SIREN14/16000. > a=fmtp:97 bitrate=24000. These three are all actually Siren7 (16kHz sample rate), but the phone is offering them three times... I've just emailed Polycom about another one of their phones doing this as well. These should all go away if you disable 'Siren14' in the Polycom phone configuration. > a=rtpmap:102 G7221/16000. > a=fmtp:102 bitrate=32000. > a=rtpmap:101 G7221/16000. > a=fmtp:101 bitrate=24000. > a=rtpmap:103 G7221/16000. > a=fmtp:103 bitrate=16000. These are also Siren7 (G.722.1, 16kHz sample rate), at various bit rates. Asterisk supports the 32kbps bit rate, so if you had 'allow=g.7221' or 'allow=siren7' in sip.conf, the first of these options would be accepted. What you want to see in the SDP for ITU G.722.1C is 'G7221/32000' with a bitrate of 48000. If you can find a phone configuration that results that SDP offer, you'll be good to go. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR reports?
Hi all, I'm struggling with figuring out how to get management information with regard to where users are within a IVR system. Does anyone have any tips on reporting process available on where users are if call to IVR is disconnected or abandoned? Thanks Magnus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR reports?
On Fri, 2009-10-23 at 22:24 +0100, Magnus Kelly wrote: > Hi all, > > I'm struggling with figuring out how to get management information with > regard to where users are within a IVR system. Does anyone have any tips > on reporting process available on where users are if call to IVR is > disconnected or abandoned? > What we do is build a string with all the options a customer has pressed and append it to the userfield in CDR at the end of the call. That way we can follow exactly where the customer was throughout the call by parsing this string. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR reports?
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Carlos Chavez > Sent: 23 October 2009 22:33 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] IVR reports? > > On Fri, 2009-10-23 at 22:24 +0100, Magnus Kelly wrote: > > Hi all, > > > > I'm struggling with figuring out how to get management information > > with regard to where users are within a IVR system. Does anyone have > > any tips on reporting process available on where users are if call to > > IVR is disconnected or abandoned? > > > > What we do is build a string with all the options a customer has > pressed and append it to the userfield in CDR at the end of the call. > That way we can follow exactly where the customer was throughout the > call by parsing this string. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001 [Magnus] Excellent prompt, does this mean it's difficult to get a real time view? In the sense of using ivr as a complex queuing system with hold advertorial and then agents at last filter. By chance do you have a snipe of your dial plan that collects the info? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to organize TFTP root directory ?
What I do is run virtual interfaces on one box, and run multiple instances of atftpd inside xinetd, each one bound to a different IP and a different root directory. Thus, my file structure looks like this - /home/phones/ /home/phones/cisco/ /home/phones/cisco/7960 (root directory for one of the atftpd instances) /home/phones/cisco/7961 (root directory for another atftpd instance) Then, with a little dhcpd.conf magic, you can easily point different sets of phones to different tftp servers, using pools that match on key Product ID (I think) strings. --Warren Selby On Fri, Oct 23, 2009 at 8:54 AM, Steve Edwards wrote: > > Steve Edwards writes: > > > >> atftpd can do PCRE substitutions to transform a requested file name > >> into something else. I've not used this facility, but I'm guessing you > >> could transform: > >> > >> SIPDefault.cnf -> cisco/SIPDefault.cnf > >> sip.cfg -> polycom/sip.cfg > >> spa841.cfg -> sipura/spa841.cfg > > > On Fri, 23 Oct 2009, Benny Amorsen wrote: > > > Cute, but all that accomplishes is renaming. I want to run a script > > which returns a different configuration based on the file name (and > > possibly the client IP address). Unfortunately there is also no > > UserAgent-header in TFTP... > > So, break out your H&S or your wallet :) > > How about adding an option to atftpd to enable "check the execute bit" on > the requested file. If the file has the execute bit, set the MAC, IP > Address, etc. as environment variables, exec the file, and return the > output to the requester? > > Then, you could write (for example) spa841.cfg as a shell script which > have access to the MAC, etc. and could run sed/replace/preprocessor/m4 and > create the "file" on the fly. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for asterix management
Have you tried accessing the IP address of your server from another computer's web browser? --Warren Selby On Fri, Oct 23, 2009 at 10:19 AM, giancarlo lombardo < gianclomba...@gmail.com> wrote: > Dear all, > I just installed asterixnow, > but no graphical interface start automaticaly neither linux nor some other, > just command line. > Shall I do something or shall I install something more ? > > Many Thanks in advance for any help. > > Giancarlo > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interfacing asterisk with a legacy PBX
On 24/10/09 00:59, Lyle Giese wrote: PATRICK KANGETHE wrote: I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, Are you planning to get rid of the legacy PBX completely? Or is Asterisk going to be a second PBX? I am going to assume you are replacing the legacy PBX. You can setup analog extensions so that you have multiple phones on each FXS channel. But they will be like a party line. If you put 6 phones on one FXS, all 6 ring at the same time, only one person can use that extension at a time. However you can add SIP phones to Asterisk and each can have their own extension instead. It just requires cat 5 cable back to a switch for each phone. Lyle Giese LCR Computer Services, Inc. Just to add my 5 cents - connecting too many phones to an FXS port can cause problems. The term is REN - ring equivalent number, and it's used to describe the maximum phones to attach to an FSX port (from memory) PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interfacing asterisk with a legacy PBX
Your question doesn't seem clear on which way you have set up already. Phone -> Asterisk -> legacy PBX -> one of the 23 extensions Or the other way? The capabilities of your specific existing PBX and asterisk need to be matched up. With an FXS you could have the other end dial with DTMF a string like *5551212#5551213# and then have asterisk use the first number as callerid. All I know about PBX's is that somebody has done similar to that. On Thu, Oct 22, 2009 at 11:48 PM, PATRICK KANGETHE wrote: > I want to interface asterisk with a legacy pbx that has around 23 > extensions through my 8 fxs card, how do i work around this? > Hint: I have already terminated 8 extensions from the legacy PBX, i was > thinking whether i can peer the extensions from the PBX i.e like 5 > extensions be peered to one extension connecting to the fxs? How can i do > this? > > Thanks in advance, > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interfacing asterisk with a legacy PBX
On Sat, 24 Oct 2009, Paul Hales wrote: > Just to add my 5 cents - connecting too many phones to an FXS port can > cause problems. The term is REN - ring equivalent number, and it's used > to describe the maximum phones to attach to an FSX port (from memory) >From my imperfect memory... REN is the Ringer Equivalency Number which approximated how many old school telephone ringers could be driven. With more modern electronics, the load from a given device is a fraction of a REN. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
