Re: [asterisk-users] Callback through extensions.conf?
On Sun, 6 Feb 2011 16:27:33 -0600, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Have you tried playing with the length of the wait? Even if you technically need 10 seconds, you could try a lower amount to see if the other priorities in that context execute... Lowering it to 5 seconds makes no difference. I also tried adding a Hangup before Wait but then the script ends before Wait. Could it be that it's just not possible to reuse a channel to dial out after it's been used to receive a call, even though it was just for a ring? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
On Mon, Feb 7, 2011 at 2:22 AM, Gilles codecompl...@free.fr wrote:
Lowering it to 5 seconds makes no difference. I also tried adding a
Hangup before Wait but then the script ends before Wait.
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring something in your configuration that I don't know
about, there's no reason that the system should just hang up the call during
the Wait() command...
Could it be that it's just not possible to reuse a channel to dial out
after it's been used to receive a call, even though it was just for a
ring?
Well, first of all, the channel (in the example dialplan and logs you posted
earlier) wouldn't even be dialing a call, it would just be responsible for
the generation of the callfile that would then cause Asterisk to spawn a
call via whatever Channel you specified
I just had a thought thoughAre you, perhaps, hanging your mobile (or
whatever) phone up after dialing into the system to trigger that context?
The reason I ask is that would make this suddenly seem more clear
Basically, try this modified version of the dialplan code:
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(SOURCE_CIDNUMBER=${
CALLERID(num)})
exten = s,n,Set(SOURCE_CIDNAME=${CALLERID(name)})
exten = s,n,NoOp(Call from ${SOURCE_CIDNAME} - ${SOURCE_CIDNUMBER})
exten = s,n,GotoIf($[${SOURCE_CIDNUMBER} = ${GSM}]?goodcid:badcid)
exten = s,n(goodcid),NoOp(CID OK)
;how to reliably detect that line is now quiet?
exten = s,n,Wait(10)
; Note From Sherwood McGowan
; By Changing the exten = s to exten = h in the section below, we
guarantee that Asterisk will execute the code IF THE CALL IS ENDED (like in
the examples given on the mailing list)
; Good Luck!
exten = h,1,NoOp(Before cp)
exten = h,n,system(cp /var/spool/asterisk/skelett.call
/var/tmp/skelett.call)
exten = h,n,NoOp(Before echo)
exten = h,n,system(echo Channel: ZAP/1/${IPPI} /var/tmp/skelett.call)
exten = h,n,NoOp(Before mv)
exten = h,n,system(mv /var/tmp/skelett.call /var/spool/asterisk/outgoing/)
Cheers!
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Re: [asterisk-users] Callback through extensions.conf?
On Mon, 7 Feb 2011 02:59:09 -0600, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your DAHDI/Zap setup... Barring something in your configuration that I don't know about, there's no reason that the system should just hang up the call during the Wait() command... The /etc/zaptel.conf and /etc/asterisk/zapata.conf are pretty basic: === cat /etc/zaptel.conf loadzone = fr defaultzone = fr fxsks=1 === cat /etc/asterisk/zapata.conf [trunkgroups] [channels] context=from_fxo switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes canpark=yes busydetect=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling = fxs_ks channel = 1 === I just had a thought thoughAre you, perhaps, hanging your mobile (or whatever) phone up after dialing into the system to trigger that context? The reason I ask is that would make this suddenly seem more clear Yes, that's exactly the idea: From my cellphone, I ring Asterisk just to notify that I want to be called back. Asterisk waits until the line stops ringing (or 10s, if there's no better way), then it dials back. ; Note From Sherwood McGowan ; By Changing the exten = s to exten = h in the section below, we guarantee that Asterisk will execute the code IF THE CALL IS ENDED (like in the examples given on the mailing list) By moving the call file part in the h extension, the code is executed. That's better :-) Problem is... Asterisk doesn't ring the phone, nothing happens: === -- Starting simple switch on 'Zap/1-1' -- Executing [s@from_fxo:1] Wait(Zap/1-1, 1) in new stack -- Executing [s@from_fxo:2] Set(Zap/1-1, SOURCE_CIDNUMBER=5551234) in new stack -- Executing [s@from_fxo:3] Set(Zap/1-1, SOURCE_CIDNAME=) in new stack -- Executing [s@from_fxo:4] NoOp(Zap/1-1, Call from - 5551234) in new stack -- Executing [s@from_fxo:5] NoOp(Zap/1-1, CID OK) in new stack -- Executing [s@from_fxo:6] Wait(Zap/1-1, 10) in new stack == Spawn extension (from_fxo, s, 6) exited non-zero on 'Zap/1-1' -- Executing [h@from_fxo:1] NoOp(Zap/1-1, Before cp) in new stack -- Executing [h@from_fxo:2] System(Zap/1-1, cp /var/spool/asterisk/skelett.call /var/tmp/skelett.call) in new stack -- Executing [h@from_fxo:3] NoOp(Zap/1-1, Before echo) in new stack -- Executing [h@from_fxo:4] System(Zap/1-1, echo Channel: ZAP/1/5551234 /var/tmp/skelett.call) in new stack -- Executing [h@from_fxo:5] NoOp(Zap/1-1, Before mv) in new stack -- Executing [h@from_fxo:6] System(Zap/1-1, mv /var/tmp/skelett.call /var/spool/asterisk/outgoing/) in new stack -- Hungup 'Zap/1-1' === To simplify further, I tried building a callfile.call manually and moving it to /var/spool/asterisk/outgoing/ ... == Channel: ZAP/1/5551234 Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 == ... but Asterisk does nothing, altough show modules says that pbx_spool.so is loaded. Weird :-/ FWIW, Asterisk runs as root, and root owns callfile.call. Maybe it's the uClinux or the Asterisk I'm using that's configured in such a way that callfiles don't work as planned. Apparently, there's no other way than callfiles to have Asterisk dial out from the dialplan? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
... but Asterisk does nothing, altough show modules says that pbx_spool.so is loaded. Weird :-/ FWIW, Asterisk runs as root, and root owns callfile.call. Maybe it's the uClinux or the Asterisk I'm using that's configured in such a way that callfiles don't work as planned. Apparently, there's no other way than callfiles to have Asterisk dial out from the dialplan? ok, first of all, it can take a little while for those spooled callfiles to be executed in Asterisk... Also, have you READ the callfile documentation? Maybe you've thought to check the modification times for the files? Are the files you've created still in the outgoing directory or are they somewhere else? If they're still in the outgoing dir, then something is causing the call files to not be executed Regarding your second inquiryare you serious? I'm not trying to be mean, but WOW... There are several ways to initate a call from the dialplan (as well as other places) The Dial command is pretty handy for thatjust an example... I'm going to have to stop assisting you at this point, it's apaprent you've not done all your homework...To not know the 'h' extension was where you could put dialplan commands to be executed after the calling channel hangs up...that's in Asterisk 101...as is the Dial command, and a few other ways of initiating calls...Hell, I'd venture to say that if I was taking a class on Asterisk configuration, callfiles would be near the END of the section on initiating calls, because they're quite often NOT used due to the presence of easier methods... Cheers bud -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any voice changer applications for Asterisk?
On 02/06/2011 05:05 PM, Sherwood McGowan wrote: AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers That seems to do exactly what the Lobstertech code does. What do people use this for? The Lobstertech one was a fun toy, but seems to be of no practical use. Changing female to male, child to adult, etc. seems pretty useful, but these modules make no attempt to perform a meaningful voice change. They would need to control the formants independent of the pitch to produce anything like a plausible voice adjustment. On Sat, Feb 5, 2011 at 9:44 PM, Steve Underwood ste...@coppice.org mailto:ste...@coppice.org wrote: On 02/06/2011 05:39 AM, Bruce B wrote: Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. It might help if you explained the kind of change you would like to make, which the lobstertech module doesn't offer. Steve Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error: Unable to create channel of type 'SIP'
Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: Unable to create channel of type 'SIP'
On Mon, Feb 7, 2011 at 7:42 AM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, I am using Trixbox 2.6.2.3, ISO install I am getting the below error in `/var/log/asterisk/full` Unable to create channel of type 'SIP' (cause 3 - No route to destination) Is there anyway to figure out which extension is causing this error ? Thank you. Best regards, Sanjay -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Now,Sanjay, don't take this personally, you just happen to be ANOTHER person who has sent an email to the list lately that just crosses my tolerance for lack of respect for the try and figure it out yourself before you ask for help mentality behind this (and most Open Source project's) mailing list. ' First solution (1.5 seconds after I read your question)...check your call detail records! ,You'll see the failed call(s)! Second solution (thought of nanoseconds after the first one)NowI want you to think just a tiny bit here...If you wanted to know if a host was reachable, what would you do?.This is the same thing, except you have a list of hosts and you need to determine WHICH one cannot be reached..You try to contact each host until you find one or more that gives you a no route to host message! ping is your friend, so is mtr, also a telnet session (over the port specified for SIP to that host in your config) could be used.. Third possible method: What level of verbosity is the server currently running at? If it's not running at 3 or higher, set verbose to at least 3. That way you will see the dialplan executions that occur just before that message. Once you see that, you'll most likely have your answer. Useful tip: Another thing you could do, set qualify=yes on your sip endpoints' configurations, since this is a no route to host issue, you'll see failure on at least one of them, which will also give you your answer. Now, I'm going to sound like a jerk, but these are all simple methods that you could/should have come up with...How many seconds did you spend thinking about the issue before you decided to ask the list for help with a question that is admittedly something you should have SOME idea regarding how to test Man, I'm starting to just get pissed...That's what, 3 questions I've seen in the last 12 or less hours where the person asking the question OBVIOUSLY doesn't want to put forth any effort on their own before asking the rest of us how to do something? Asterisk Documentation is your friend! UNDERSTANDING at least 25% of how VoIP works is handy! GOOGLE is your friend! And in the name of any and all things/beings that you guys find to be holy, put forth some damned effort before asking everyone else to do the work for you Finally, if you HAVE put forth effort, LET US KNOW!!! It lessens the chance of you getting flamed by some guy who's been working for over 40 hours STRAIGHT and is just tired of seeing email after email after email containing questions that have been answered hundreds of times on the list and there are readily available answers via documentation and/or a little friggin googling.. That's it...I'm going back to barely reading the list...Every time I try to start reading it on a fairly often basis (in the hopes of being able to help people with continuing issues AFTER putting some damn effort towards the problem), I start seeing that 75-80% of new requests have 0-5% effort put forth into trying to fix it themselves, and this includes basic stuff like RTFM! Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: ok, first of all, it can take a little while for those spooled callfiles to be executed in Asterisk... Thanks for your help. The same callfile works fine in Ubuntu, but not at that appliance. Since I can dial through the FXO, it doesn't seem to be a Zaptel issue either. I'll investigate further, and find a work-around if the appliance just doesn't support this feature for some reason. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About maxlen parameter in queues
Dear list, I want to avoid sending calls to a queue when it is full. From the fact that 'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8 Thanks, Elder Arohuanca Lagos t. 992728100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any voice changer applications for Asterisk?
On Mon, Feb 7, 2011 at 8:39 AM, Steve Underwood ste...@coppice.org wrote: On 02/06/2011 05:05 PM, Sherwood McGowan wrote: AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers That seems to do exactly what the Lobstertech code does. What do people use this for? The Lobstertech one was a fun toy, but seems to be of no practical use. Changing female to male, child to adult, etc. seems pretty useful, but these modules make no attempt to perform a meaningful voice change. They would need to control the formants independent of the pitch to produce anything like a plausible voice adjustment. Thanks for the clarification. I got to agree that it's not of practical use. I was hoping there is a way around the echo and long delay that is generated. I guess not yet. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About maxlen parameter in queues
This might be what you're looking for http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action +QueueStatus Ish On Mon, 2011-02-07 at 10:44 -0500, Daniel - Asterisk wrote: Hi Danny, Could you please let me know what function do I use to get if the queue is full? Elder On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas da...@debsinc.com wrote: __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel - Asterisk Sent: Monday, February 07, 2011 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] About maxlen parameter in queues Dear list, I want to avoid sending calls to a queue when it is full. From the fact that 'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8 Thanks, Elder Arohuanca Lagos t. 992728100 This is a bit “hackish”, but why don’t you just make a context that uses AGI to query the queue and only let the call proceed if not full? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] downgrade libpri
I`m currently having aleatory call drops through a PRI and so I want to upgrade libpri. In case I have errors I will want to downgrade libpri, that`s the reason why I asked. Asterisk version: 1.6.2.13 Current Libpri: libpri-1.4.10.2 Libpri to be upgraded to: libpri-1.4.11.5 Any help will be thanked! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a duration limit be specified in spool call file?
Bruce B wrote: Oh, no problem. I do understand that there are a bunch of you guys who would like to jump on a topic with smart-/aleck remarks and /act as moderators of the list. This is nothing new to me and I understand that you totally defy the whole purpose of the mailing list. Keeping in mind that you guys are continuously the ones who try to be smart-asses, no offence is taken at all. Please continue with the attitude. I understand that there are certain males with higher estrogen levels than needed who can PMS from time to time. Klinefelter's does happen in 1 in 1000 males. I am sure you can search that up on the magical Google search engine you pointed me to. Cheers bud, Sorry, feeling a bit honest today; had to blur it out Some of these same aleck's are so intent on commenting on someone's perceived bad behavior they can't be bothered to do a proper trim, even of the list footers that serve no purpose other than to take up space on subsequent messages. The self appointed list policeman is a chronic offender who is also quite vocal on what style of posting should or should not be done the good news is that many are more than willing to help in some way, regardless of the few. John Novack On Sun, Feb 6, 2011 at 9:42 PM, Sherwood McGowan sherwood.mcgo...@gmail.com mailto:sherwood.mcgo...@gmail.com wrote: On Sun, Feb 6, 2011 at 5:35 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Sun, 6 Feb 2011, Bruce B wrote: Can you you please explain the Local Channel concept. I am not sure what should be the Channel line: Channel: xxx/yyy/ Gosh. This was the first result returned by googling 'asterisk local channel.' http://www.voip-info.org/wiki/view/Asterisk+local+channels While there is a lot of out of date crap out there, www.voip-info.org http://www.voip-info.org is still a valuable resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 snark OhSteve...it gets better...Did you know that apparently there's documentation that accompanies Asterisk? Apparently, if you go to the Asterisk website and click the link for Documentation.../snark The Official Asterisk Wiki, or /The Official Asterisk Documentation/: https://wiki.asterisk.org/wiki/display/AST There's a section on Asterisk's Channel Drivers, and a subsection on Local Channels: https://wiki.asterisk.org/wiki/display/AST/Local+Channel Sorry Bruce B, but ya kinda walked into that one by asking the mailing list to explain the Local channel concept instead of doing a quick search on google or even checking the documentation... Even the pre 1.8 versions of Asterisk had a doc subdirectory in the source tree, and IIRC there was a text file called something like Local Channels.txt or Local Channels.readme, something like that. Please, do some of the light lifting, or Steve Edwards will get you...and if he doesn't, rest assured that someone will (myself, when feeling particularly grumpy, included)Most of us are pretty helpful on this list, but we don't like repeating ourselves or having to post links to easily googled resources... Cheers bud! Sherwood McGowan ...feelin' a little grumpy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6
Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend auth=md5 accountcode=user1 notransfer=yes context=context1 host=10.0.0.250 username=user1 secret=secret1 disallow=all allow=alaw [user2] type=friend auth=md5 accountcode=user2 notransfer=yes context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/24 username=user2 secret= disallow=all allow=alaw If a call comes in from 10.0.0.250, using username user2 and with no password, then it is correctly authenticated against the [user2] section. Accountcode is set to user2 Context is set to context2 and the call mostly proceeds correctly, BUT the source channel name is set to IAX2/user1-, which is then seen both in the dialplan debug output, and in the CDR. I would expect the channel name to reflect the section name that was used to authenticate the call ie. IAX2/user2-; I specifically put a password onto [user1] so there is no possibility that the call is authenticating there. Am I missing something? Or is this a bug? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) Thanks Steve for the help. I launched * with asterisk -d -d -d -n -v -v -v -c, and ran module show to check that pbx_spool.so is loaded: = *CLI module show like pbx_spool.so Module Description Use Count pbx_spool.so Outgoing Spool Support 0 1 modules loaded = Next, I moved the following callfile to /var/spool/asterisk/outgoing: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Asterisk runs as root, and owns this file as well. Unfortunately, nothing shows up in the console, the xlite extension isn't called, even after waiting for a few minutes. Could it be that pbx_spool.so isn't really loaded, or is Asterisk somehow configured to ignore callfiles? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MP3 Crashing Asterisk
On Saturday 05 Feb 2011, Timothy Smith wrote: On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Can you listen to an mp3 file through the Asterisk server's own sound card (if it has one; if not, use the -w option to write to a .wav file, and test that by copying it to another machine which has a sound card), by invoking mpg123 from the command line? Unfortunately, I cannot as the server is in a remote location. I also have to read about crash dumps to establish which file exactly cuases the crash. I have too much debugging but I usually see [Feb 5 08:15:51] WARNING[4895] mp3/interface.c: Junk at the beginning of frame 49443303 or [Feb 5 02:14:05] WARNING[7447]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304 just before the crash. OK. If you've no physical access to the Asterisk server then use something like $ mpg123 -w filename.wav filename.mp3 to convert an MP3 file to a .wav file, and see if that works -- it should take noticeably less time than the length of the sound file. If that seems to work without crashing, then copy the resulting .wav file to some machine with a sound card and make sure it plays OK. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] downgrade libpri
On Mon, Feb 07, 2011 at 02:13:32PM -0200, Agustina Berretta wrote: I`m currently having aleatory call drops through a PRI and so I want to upgrade libpri. In case I have errors I will want to downgrade libpri, that`s the reason why I asked. Upgrading? Downgrading? Asterisk version: 1.6.2.13 Current Libpri: libpri-1.4.10.2 Libpri to be upgraded to: libpri-1.4.11.5 Installed from packages? From source? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
On Mon, Feb 7, 2011 at 10:46 AM, Gilles codecompl...@free.fr wrote: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Just a thought... Did you originally generate this callfile on the linux box itself, or did you create it first on a windows box and then move it over to the linux box? There's something to do with the way Windows text editor does line breaks that linux doesn't like, and asterisk expects line breaks in certain spots for it to work properly. If that isn't the case - maybe look at using a local channel instead of SIP/xlite to setup the call? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
Real quick, please respond to my question about where the callfile ends up after a few minutes, as well as the modification time and the permissions on the file ;-) These are good bits to know On Mon, Feb 7, 2011 at 10:46 AM, Gilles codecompl...@free.fr wrote: On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) Thanks Steve for the help. I launched * with asterisk -d -d -d -n -v -v -v -c, and ran module show to check that pbx_spool.so is loaded: = *CLI module show like pbx_spool.so Module Description Use Count pbx_spool.so Outgoing Spool Support 0 1 modules loaded = Next, I moved the following callfile to /var/spool/asterisk/outgoing: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Asterisk runs as root, and owns this file as well. Unfortunately, nothing shows up in the console, the xlite extension isn't called, even after waiting for a few minutes. Could it be that pbx_spool.so isn't really loaded, or is Asterisk somehow configured to ignore callfiles? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a duration limit be specified in spool call file?
Bruce, it's all good my man Hey Novack...while I'm sure that both of you are quite pleased with your own witty smart-aleck remarks, let me first say, in the past I've sat back (for quite some time, seeing as how I've been on this list for close to what 5 or 6 years under varied emails throughout that time) and watched as others played policeman and still others continued to defeat the purpose of...wait for itthe ARCHIVES of the mailing list and the DOCUMENTATION...and said nary a word..In fact, I daresay that I was helpful...Hell, here's one for those who think I'm just ALWAYS an asshole...you know how asterisk moved away from using macros and migrated to gosubs? yeah...guess who it was that not only reported the issue and worked with Murf to figure out what was causing the crashes (that we now know is due to too many recursive macro calls, something like 7 I believe did the trick) On Mon, Feb 7, 2011 at 10:29 AM, John Novack jnov...@stromberg-carlson.orgwrote: Bruce B wrote: Oh, no problem. I do understand that there are a bunch of you guys who would like to jump on a topic with smart-*aleck remarks and *act as moderators of the list. This is nothing new to me and I understand that you totally defy the whole purpose of the mailing list. Keeping in mind that you guys are continuously the ones who try to be smart-asses, no offence is taken at all. Please continue with the attitude. I understand that there are certain males with higher estrogen levels than needed who can PMS from time to time. Klinefelter's does happen in 1 in 1000 males. I am sure you can search that up on the magical Google search engine you pointed me to. of Cheers bud, Sorry, feeling a bit honest today; had to blur it out Some of these same aleck's are so intent on commenting on someone's perceived bad behavior they can't be bothered to do a proper trim, even of the list footers that serve no purpose other than to take up space on subsequent messages. The self appointed list policeman is a chronic offender who is also quite vocal on what style of posting should or should not be done the good news is that many are more than willing to help in some way, regardless of the few. John Novack On Sun, Feb 6, 2011 at 9:42 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Sun, Feb 6, 2011 at 5:35 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 6 Feb 2011, Bruce B wrote: Can you you please explain the Local Channel concept. I am not sure what should be the Channel line: Channel: xxx/yyy/ Gosh. This was the first result returned by googling 'asterisk local channel.' http://www.voip-info.org/wiki/view/Asterisk+local+channels While there is a lot of out of date crap out there, www.voip-info.org is still a valuable resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 snark OhSteve...it gets better...Did you know that apparently there's documentation that accompanies Asterisk? Apparently, if you go to the Asterisk website and click the link for Documentation.../snark The Official Asterisk Wiki, or *The Official Asterisk Documentation*: https://wiki.asterisk.org/wiki/display/AST There's a section on Asterisk's Channel Drivers, and a subsection on Local Channels: https://wiki.asterisk.org/wiki/display/AST/Local+Channel Sorry Bruce B, but ya kinda walked into that one by asking the mailing list to explain the Local channel concept instead of doing a quick search on google or even checking the documentation... Even the pre 1.8 versions of Asterisk had a doc subdirectory in the source tree, and IIRC there was a text file called something like Local Channels.txt or Local Channels.readme, something like that. Please, do some of the light lifting, or Steve Edwards will get you...and if he doesn't, rest assured that someone will (myself, when feeling particularly grumpy, included)Most of us are pretty helpful on this list, but we don't like repeating ourselves or having to post links to easily googled resources... Cheers bud! Sherwood McGowan ...feelin' a little grumpy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Can a duration limit be specified in spool call file?
oh and didn't you guys already have your little histrionics sessin about trimming the goddamned emails, mailing list etiquette about top posting versus bottom, etc../.. My complaint is not something as trivial as where one should reply in a mailing list email, or if one should trim emails every time he replies to the list... The reason why I have my occasional bitch session at a few emails is not that I want to be a policeman..it's twofold... 1. I don't let my son pull that shit when I know he can at least partially figure it out...He'll learn it 4-5 times slower if he's just handed answers to problems...Look at it this way, at least I only reply to emails playing mailing list policeman (gotta love how that was thrown in just before the gentlemen played his own version of enforcer of etiquette) occasionally...my kid hears it all the time 2. I can only take so many years of doing my best to NOT be one of those people who get a ticket from the list police, learning 98% of my trade by experimentation and research, until I start feeling like it's fairly rude to see the amount of please give me the information without me having to put in much work that this list has become...go read the archives...it's getting exponentially worse... Let me close by saying, Bruce, I did go overboard on your email. I apologize. You'll find that I don't do THAT often either, but I am a man, and I'm not playing the it's the internet so my balls are 250% bigger game. Novack, I'll close by saying, have a nice day...and I'll keep trimming occassionally, as I always have...what can I say, I'm not THAT concerned with space...I have plenty of screens...etc...etc... To the rest of you, I sincerely hope you didn't waste your time watching this childish set of insults and rants... *grabs a bucket...tosses water on the laptop*** flame off, here's hoping we can get back to work On Mon, Feb 7, 2011 at 11:12 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Bruce, it's all good my man Hey Novack...while I'm sure that both of you are quite pleased with your own witty smart-aleck remarks, let me first say, in the past I've sat back (for quite some time, seeing as how I've been on this list for close to what 5 or 6 years under varied emails throughout that time) and watched as others played policeman and still others continued to defeat the purpose of...wait for itthe ARCHIVES of the mailing list and the DOCUMENTATION...and said nary a word..In fact, I daresay that I was helpful...Hell, here's one for those who think I'm just ALWAYS an asshole...you know how asterisk moved away from using macros and migrated to gosubs? yeah...guess who it was that not only reported the issue and worked with Murf to figure out what was causing the crashes (that we now know is due to too many recursive macro calls, something like 7 I believe did the trick) On Mon, Feb 7, 2011 at 10:29 AM, John Novack jnov...@stromberg-carlson.org wrote: Bruce B wrote: Oh, no problem. I do understand that there are a bunch of you guys who would like to jump on a topic with smart-*aleck remarks and *act as moderators of the list. This is nothing new to me and I understand that you totally defy the whole purpose of the mailing list. Keeping in mind that you guys are continuously the ones who try to be smart-asses, no offence is taken at all. Please continue with the attitude. I understand that there are certain males with higher estrogen levels than needed who can PMS from time to time. Klinefelter's does happen in 1 in 1000 males. I am sure you can search that up on the magical Google search engine you pointed me to. of Cheers bud, Sorry, feeling a bit honest today; had to blur it out Some of these same aleck's are so intent on commenting on someone's perceived bad behavior they can't be bothered to do a proper trim, even of the list footers that serve no purpose other than to take up space on subsequent messages. The self appointed list policeman is a chronic offender who is also quite vocal on what style of posting should or should not be done the good news is that many are more than willing to help in some way, regardless of the few. John Novack On Sun, Feb 6, 2011 at 9:42 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Sun, Feb 6, 2011 at 5:35 PM, Steve Edwards asterisk.org@ sedwards.com wrote: On Sun, 6 Feb 2011, Bruce B wrote: Can you you please explain the Local Channel concept. I am not sure what should be the Channel line: Channel: xxx/yyy/ Gosh. This was the first result returned by googling 'asterisk local channel.' http://www.voip-info.org/wiki/view/Asterisk+local+channels While there is a lot of out of date crap out there, www.voip-info.orgis still a valuable resource. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline
[asterisk-users] remote bridging
Hi List, Quick question: I am using asterisk 1.8.1 - how do I detect whether a native (remote) bridge is being used between 2 SIP peers? It is not obvious to me from the console logs. Thanks, Ondrej -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
On 02/07/2011 11:46 AM, Gilles wrote: snip Asterisk runs as root, and owns this file as well. Have you tried setting the permissions of this file to world readable, to ensure that any user can read it and eliminate potential permissions problems? Worth a shot. While you're at it, output from the ps command that shows the output, command line, and header for asterisk will help, too. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remote bridging
Actually mate, I'm about to start a run of testing on a project that actually applies...if I figure it out shortly, I'll respond :D On Mon, Feb 7, 2011 at 11:49 AM, Ondrej Valousek webs...@s3group.cz wrote: Hi List, Quick question: I am using asterisk 1.8.1 - how do I detect whether a native (remote) bridge is being used between 2 SIP peers? It is not obvious to me from the console logs. Thanks, Ondrej -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec negotiation
Hi List, I am using asterisk 1.8.1. and I want to avoid transcoding when 2 SIP peers calling each other: A (g722, alaw) calls B (alaw,ulaw) via asterisk. My setup: allow=g722,alaw preferred_codec_only=no Note that when B calls A, codec alaw is used on both ends, fine, but it does not seem to work the reverse way (A is using g722, B is using alaw, asterisk is doing transcoding). Is it possible? Thanks, Ondrej -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a duration limit be specified in spool call file?
On Mon, Feb 7, 2011 at 12:40 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: oh and didn't you guys already have your little histrionics sessin about trimming the goddamned emails, mailing list etiquette about top posting versus bottom, etc../.. My complaint is not something as trivial as where one should reply in a mailing list email, or if one should trim emails every time he replies to the list... The reason why I have my occasional bitch session at a few emails is not that I want to be a policeman..it's twofold... 1. I don't let my son pull that shit when I know he can at least partially figure it out...He'll learn it 4-5 times slower if he's just handed answers to problems...Look at it this way, at least I only reply to emails playing mailing list policeman (gotta love how that was thrown in just before the gentlemen played his own version of enforcer of etiquette) occasionally...my kid hears it all the time 2. I can only take so many years of doing my best to NOT be one of those people who get a ticket from the list police, learning 98% of my trade by experimentation and research, until I start feeling like it's fairly rude to see the amount of please give me the information without me having to put in much work that this list has become...go read the archives...it's getting exponentially worse... Let me close by saying, Bruce, I did go overboard on your email. I apologize. You'll find that I don't do THAT often either, but I am a man, and I'm not playing the it's the internet so my balls are 250% bigger game. Novack, I'll close by saying, have a nice day...and I'll keep trimming occassionally, as I always have...what can I say, I'm not THAT concerned with space...I have plenty of screens...etc...etc... To the rest of you, I sincerely hope you didn't waste your time watching this childish set of insults and rants... *grabs a bucket...tosses water on the laptop*** flame off, here's hoping we can get back to work Good to know you are not short of space :-) I hope everyone else upgrades as well. I wouldn't have minded you saying, *This topic is fully covered in /Doc folder and in asterisk Wiki. Reference URL: .* in one or two lines that was relevant. To go on a lecture. I may still not agree with your points but apology accepted and no hurt feelings here either. I hope that Asterisk mailing list becomes a less hostile list day by day. Cheers (for real) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
Asterisk runs as root but what about the bash script or the php file that creates the file? Maybe comment the mv command and check the file permissions by *ls -la call-filename.call* to be sure. *chown root.root call-filename* (if root is really the user running Asterisk) and then the mv command line should do the trick. If you are sure that permissions are not the problem and you have archive set to yes then you can browse the */var/spoo/asterisk/outgoing_done* folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce On Mon, Feb 7, 2011 at 11:46 AM, Gilles codecompl...@free.fr wrote: On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: sudo /usr/sbin/asterisk -d -d -d -n -v -v -v Oops. A '-c' should be in there :) Thanks Steve for the help. I launched * with asterisk -d -d -d -n -v -v -v -c, and ran module show to check that pbx_spool.so is loaded: = *CLI module show like pbx_spool.so Module Description Use Count pbx_spool.so Outgoing Spool Support 0 1 modules loaded = Next, I moved the following callfile to /var/spool/asterisk/outgoing: = #callfileSIP.call Channel: SIP/xlite Context: callback-dialtone-auth Extension: s Priority: 1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 = Asterisk runs as root, and owns this file as well. Unfortunately, nothing shows up in the console, the xlite extension isn't called, even after waiting for a few minutes. Could it be that pbx_spool.so isn't really loaded, or is Asterisk somehow configured to ignore callfiles? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
*** ever so politely snipping *** If you are sure that permissions are not the problem and you have archive set to yes then you can browse the */var/spoo/asterisk/outgoing_done*folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce The archived file, if I recall correctly, will be appended with failed or something similar...I'd check the wiki -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback through extensions.conf?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, February 07, 2011 12:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callback through extensions.conf? *** ever so politely snipping *** If you are sure that permissions are not the problem and you have archive set to yes then you can browse the /var/spoo/asterisk/outgoing_done folder to see if the call file is transferred there or not. The file should contain some info to help you and it's existence also means that somehow you are not seeing the call through your CLI as it's processed. However I doubt this is happening. -Bruce The archived file, if I recall correctly, will be appended with failed or something similar...I'd check the wiki In my (1.4.X) experience, the file just stays in /var/spool/asterisk/outgoing and gets little tags added until you get the problem resolved or delete the file. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: SwitchVox Mailing List?
Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 http://forums.digium.com/viewtopic.php?f=38t=77031sid=4adb81c464701e0039d e21a300aa273f t=77031sid=4adb81c464701e0039de21a300aa273f William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: SwitchVox Mailing List?
Try these - SwitchVox SMB and SOHO http://forums.digium.com/viewforum.php?f=38 http://forums.digium.com/viewforum.php?f=38sid=e78c5fda089b88d8e1617d0c548 d8f8d sid=e78c5fda089b88d8e1617d0c548d8f8d SwitchVox Free Editiion Support http://forums.digium.com/viewforum.php?f=19 http://forums.digium.com/viewforum.php?f=19sid=e78c5fda089b88d8e1617d0c548 d8f8d sid=e78c5fda089b88d8e1617d0c548d8f8d Hope this is useful _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Monday, February 07, 2011 12:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] OT: SwitchVox Mailing List? Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 http://forums.digium.com/viewtopic.php?f=38t=77031sid=4adb81c464701e0039d e21a300aa273f t=77031sid=4adb81c464701e0039de21a300aa273f William Stillwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a duration limit be specified in spool call file?
Bruce, All in all, I don't think it's that hostile, it just goes through cycles...maybe a good number of us may indeed have estrogen issues and it's the moon, who knows ;-) LOL Cheers (and I always mean it, seriously :D ) Sherwood McGowan Yes, THAT Mick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple inbound calls from same sip trunk
Hi everybody, I have two toll free numbers pointed to my asterisk server. My toll free number provider gave me two 7 digit dnis numbers. Both numbers land in the extensions. How to make the softphone (xlite) know that the call has landed through which number? I think the differentiating stuff is the dnis numbers. Is there any way, where I can notify the softphone in regard with the dnis number? Any help will be highly appreciated. Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple inbound calls from same sip trunk
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohan Shahi
Sent: Monday, February 07, 2011 12:57 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] multiple inbound calls from same sip trunk
Hi everybody,
I have two toll free numbers pointed to my asterisk server. My toll free
number provider gave me two 7 digit dnis numbers. Both numbers land in the
extensions.
How to make the softphone (xlite) know that the call has landed through
which number? I think the differentiating stuff is the dnis numbers. Is
there any way, where I can notify the softphone in regard with the dnis
number?
Any help will be highly appreciated.
Thank you
If the value isn't in ${EXTEN} or ${CALLERID(num)} you'll need to query SIP
headers, but I'd be shocked if those two options don't do it for you.
--
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Re: [asterisk-users] Callback through extensions.conf?
In my (1.4.X) experience, the file just stays in /var/spool/asterisk/outgoing and gets “little tags” added until you get the problem resolved or delete the file. That is absolutely true if the file is not processed. I guess he can again do a ls -la in that folder to check permissions for the file not processed. -Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] standalone NOTIFY message handling for Asterisk
Hi Group, Do you think this has been fixed or it's still not supported with standalone NOTIFY? Your help will be highly appreciated! Regards, Felton - Original Message From: Feng Xu felto...@yahoo.com To: asterisk-users@lists.digium.com Sent: Fri, 4 February, 2011 1:43:39 PM Subject: [asterisk-users] standalone NOTIFY message handling for Asterisk Hi, I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS), currently when VMS send NOTIFY message (standalone NOTIFY, no previous SUBSCRIBE for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug log it indicates as the following: [Feb 4 13:27:06] DEBUG[8353] chan_sip.c: Invalid SIP message - rejected , no callid, len 771 I have googled around regarding to this topic, I found similar issues reported, but that's all the old version back to 2004 around. So I would like to confirm, does Asterisk support standalone NOTIFY message for message waiting indicator regarding to voice mail. If yes, what's the configurations should be applied? My current configuration with Asterisk: sip.conf = [vms] type = peer -- tried both friend and peer host = a.b.c.d -- the VMS IP address unsolicited_mailbox = yes fromuser = lab_vms . [user1] mailbox = user1@SIP_Remote The NOTIFY message received: == NOTIFY sip:+user1@a.b.c.d:5060 SIP/2.0^M Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK-781458841-7-21411^M CSeq: 1 NOTIFY^M From: sip:lab_vms@a.b.c.d:5060;tag=1426753238-7761-21411^M To: sip:+user1@a.b.c.d:5060^M Call-ID: 1231979610-5-21411-CmvtCallId^M Route: sip:a.b.c.d:5060;transport=udp;lr^M Event: message-summary^M Accept: application/sdp,application/media_control+xml,message/sipfrag^M Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,INFO,PRACK,UPDATE^M Contact: sip:a.b.c.d:5060;transport=udp^M Max-Forwards: 70^M Supported: 100rel,timer,histinfo^M Subscription-State: active^M MIME-Version: 1.0^M Content-Type: application/simple-message-summary^M Content-Length: 40^M ^M Messages-Waiting: yes^M None: 5/0 (0/0)^M Thanks a lot for your help! Felton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] standalone NOTIFY message handling for Asterisk
Hi, Here is a solution to your problem. By default asterisk send all OPTION messages to default context in dialplan regardless on peers's context. You will get 200 OK reply to your option packet if you add following lines to dialplan, [default] exten = _X.,1,NoOp() exten = _X.,n,Hangup() exten = h,1.NoP() Regards, Faisal -Original Message- From: Feng Xu felto...@yahoo.com Sent: Monday, February 7, 2011 11:55pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] standalone NOTIFY message handling for Asterisk Hi Group, Do you think this has been fixed or it's still not supported with standalone NOTIFY? Your help will be highly appreciated! Regards, Felton - Original Message From: Feng Xu felto...@yahoo.com To: asterisk-users@lists.digium.com Sent: Fri, 4 February, 2011 1:43:39 PM Subject: [asterisk-users] standalone NOTIFY message handling for Asterisk Hi, I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS), currently when VMS send NOTIFY message (standalone NOTIFY, no previous SUBSCRIBE for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug log it indicates as the following: [Feb 4 13:27:06] DEBUG[8353] chan_sip.c: Invalid SIP message - rejected , no callid, len 771 I have googled around regarding to this topic, I found similar issues reported, but that's all the old version back to 2004 around. So I would like to confirm, does Asterisk support standalone NOTIFY message for message waiting indicator regarding to voice mail. If yes, what's the configurations should be applied? My current configuration with Asterisk: sip.conf = [vms] type = peer -- tried both friend and peer host = a.b.c.d -- the VMS IP address unsolicited_mailbox = yes fromuser = lab_vms . [user1] mailbox = user1@SIP_Remote The NOTIFY message received: == NOTIFY sip:+user1@a.b.c.d:5060 SIP/2.0^M Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK-781458841-7-21411^M CSeq: 1 NOTIFY^M From: sip:lab_vms@a.b.c.d:5060;tag=1426753238-7761-21411^M To: sip:+user1@a.b.c.d:5060^M Call-ID: 1231979610-5-21411-CmvtCallId^M Route: sip:a.b.c.d:5060;transport=udp;lr^M Event: message-summary^M Accept: application/sdp,application/media_control+xml,message/sipfrag^M Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,INFO,PRACK,UPDATE^M Contact: sip:a.b.c.d:5060;transport=udp^M Max-Forwards: 70^M Supported: 100rel,timer,histinfo^M Subscription-State: active^M MIME-Version: 1.0^M Content-Type: application/simple-message-summary^M Content-Length: 40^M ^M Messages-Waiting: yes^M None: 5/0 (0/0)^M Thanks a lot for your help! Felton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can a duration limit be specified in spool call file?
Hi, If you need full control on both legs of call you can redirect Leg-1 to your dialplan as [mailto:Local/your-extension@your-context/n] Channel: Local/your-extension@your-context/n and from there you control the Leg-1 using dial-plan or AGI as you like while Leg is normally comes to dialplan and totally in controll. Regards, Faisal scussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can a duration limit be specified in spool call file? Bruce, All in all, I don't think it's that hostile, it just goes through cycles...maybe a good number of us may indeed have estrogen issues and it's the moon, who knows ;-) LOL Cheers (and I always mean it, seriously :D ) Sherwood McGowan Yes, THAT Mick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec negotiation
Hi, If you will send call without answering on asterisk and have directrtpsetup=yes in sip.conf codec negociation will always be between UAs so any matched codec will work fine. If you are answering call on asterisk then dialing it out to next UA then you need to add canreinvite=yes for both UAs. Regards, Faisal P peers calling each other: A (g722, alaw) calls B (alaw,ulaw) via asterisk. My setup: allow=g722,alaw preferred_codec_only=no Note that when B calls A, codec alaw is used on both ends, fine, but it does not seem to work the reverse way (A is using g722, B is using alaw, asterisk is doing transcoding). Is it possible? Thanks, Ondrej -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
