IRC log for #asterisk on 20090603

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01:53.30jayteehttp://tinyurl.com/p3dwbo
01:53.35jayteeyum!
01:53.42jaytee:-)
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01:55.09Qwellapt-get!
01:55.25jayteehehe
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02:02.30uluatu_people, is this possible to devicestate not report the correct state of some channel after an attendant transfer?
02:11.45[TK]D-Fenderjaytee: Engrish FAIL
02:12.23jaytee:-)
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04:15.14jnfullerdoes anyone know if there are any good forums or chat rooms for digium's free fax for asterisk>
04:17.28jnfulleror, failing that... if you buy a channel do you get access to any better docs/support?
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04:31.02lanningquick SIP question: To: format is <user>@<server>/<did> correct?
04:31.45lanningthis is the sip packet to: field...
04:32.39[TK]D-Fenderlanning: Not TO a user, FROM a user..
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04:42.08jnfullerman do I hate peering with an acme-packet sbc
04:42.54jnfullerone screwup in a regex and all the calls die with 404
04:43.22jnfullerbut no, everything's fine on their end
04:43.26jnfuller*sigh*
04:43.43jnfullerescape + often?
04:43.52jnfullersorry, I'll keep my rant to myself
04:44.01jnfullerjust needed to vent
04:44.10carrartell us how you really feel
04:44.49[TK]D-Fendercarrar: I would, but its illegal in 17 states.
04:44.50[TK]D-Fendercarrar : But she is SO worth it....
04:44.54InfoMoMocarrar: well said :D
04:45.11carrarheh
04:45.29[TK]D-Fendercarrar: And the doctors tell me feeling should return there within a week
04:45.54carrarThats the good times!
04:46.57[TK]D-FenderWE ARE FOR GOOD TIMES!
04:47.33*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
04:50.49jnfullerlanning, are you looking for rfc 3261 for uri's or how to place an outbound call in asterisk?
04:51.34*** join/#asterisk Jacke (i=jacke@valhalla.bofh.org)
04:51.54lanningthe uri directly in the sip packet...
04:52.19jnfullerthe uri follows 3261
04:53.56jnfullerhttp://www.ietf.org/rfc/rfc3261.txt
04:54.32lanningas reported in sip set debug
04:54.32lanningya, just been reading up.
04:54.32lanninglooks like the provider is handing me DIDs as separate users.
04:55.50jnfullerthere's really no such concept as did in sip, user@host
04:56.50florzwhat's DID then, if there is no such concept in SIP?!
04:57.23jnfullerthe did is arbitrary, it's just a text field for a user
04:57.41jnfullerso it's up to the ua to decide if it is valid
04:58.04*** join/#asterisk mchou_ (n=mchou@unaffiliated/mchou)
04:58.22florzso you say that the DID is arbitrary and therefore is not a DID?!?
04:58.33carrarNo, there is no spoon
04:58.47[TK]D-Fenderhordes the cutlery drawer
04:58.57[TK]D-FenderMY PRECIOUS!!!!!!!!!
04:58.58jnfulleryes for example a provider sends foo@bar.baz
04:59.11jnfullerfoo is just a username and could be anything
04:59.27jnfullerit's up to the downstream server to react
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04:59.31carrarShould have use mailer daemon, that guy is always emailing me
04:59.38[TK]D-Fender"sends"?  Could someone be more vague please?
05:00.17jnfullerno more vague than using a PRI term like did with sip ;)
05:00.34carrarPRI over SIP
05:00.48carrarPRIoSIP
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05:01.11k-man_is ther Siemens C470 ip phone any good?
05:02.38lanningtrying to get it to not use extension s
05:02.59florzjnfuller: Why is "DID" a "PRI term"? Is "phone call" a "PRI term", too?
05:03.33[TK]D-Fenderand DID is not a PRI term...
05:03.48[TK]D-FenderThere are higher protocols at work.
05:03.57florzbut phone call is? =:-)
05:04.46[TK]D-Fenderlanning: If you are receiving a call targeting "s" its because you didn't tell then any other number to dial when sending you calls
05:04.51jnfullerall I meant was that from a sip perspective it's user@host
05:04.53[TK]D-Fenderthem*
05:05.24florzjnfuller: _what_ is "user@host", and what does that have to do with the definition of "DID"?
05:05.52jnfullerWell, what's your definition of did ?
05:06.09florzthat was kindof the question I was asking you, I guess =:-)
05:06.22lanninghttp://pastebin.com/d72e67f5
05:06.23jnfullerTo me, DID is an assigned number
05:06.43jnfullerone coming from a provider, really it's the same as an assigned sip user
05:07.01jnfullerlogically, not at a protocol level
05:07.33[TK]D-Fenderlanning: INVITE sip:s@10.0.1.254 SIP/2.0 <- because you didn't tell them where to send to
05:07.48florzhow about "telling the receiver of the signalling some information that allows it to distinguish different application level destination addresses"?
05:08.07lanninggreat, I was looking at the from/to fields, not the invite line... :)
05:08.13jnfullerright, which leads me back to sip is luser@host
05:08.41florzyeah?
05:08.56jnfullerand back to rfc 3261, which was the original ask
05:08.56florzYou mean, like, in SIP there is no way to accomplish that?
05:10.11*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83)
05:10.25jnfullerI'm lost now. I'm saying I'm jnfuller@blah.baz and you're florz@maz.boo and if you want to call me you address me as jnfuller@blah.baz... technically, that's my did
05:10.49jnfullerfunctionally, but in sip there's no such thing as did, just uri
05:10.59carrarDirect Inward Dialing
05:10.59jnfullerare we arguing the same side?
05:11.20[TK]D-FenderLESS FILLING!!!
05:11.26carrarTASTE GREAT
05:11.31[TK]D-FenderLESS FILLING!!!!!!!!
05:11.31florzjnfuller: I guess what you are calling "DID" is commonly called a "telephone number"
05:11.38jnfulleror sip uri
05:11.38carrarTASTE GREATER!!
05:11.40[TK]D-Fendergrabs his ClueBat (tm)
05:11.54[TK]D-Fendercarrar: Might makes right, and I am very VERY right! :p
05:11.57jnfullerthere's really no such thing as a telephone number in sip
05:12.01florzjnfuller: no, a SIP URI is called a SIP URI, usually
05:12.20carrarTK, You wanna call that a DID?
05:12.23carrarLets call that DID
05:12.34carrarI'm gonna call everything a DID
05:12.48carrarFARKLOR
05:12.52[TK]D-Fendergoes off to confuse Wikipedia users some more.
05:13.11florzjnfuller: BTW, jnfuller@blah.baz is syntactically not a SIP URI
05:13.24jnfulleryeah yeah, it was an abstraction
05:13.42florzjnfuller: 07:07 < florz> how about "telling the receiver of the signalling some information that allows it to distinguish different application level destination addresses"?
05:14.30[TK]D-Fendercarrar: did you do what you did when you said you did get the did I did tell you you did need to get?
05:15.03carrarI did
05:15.08florzthat concept of DID can be implemented with SIP and SIP URIs just as well as with ISDN protocols and telephone numbers
05:15.10[TK]D-Fender\o/
05:15.37jnfullerconceptually yes
05:15.39florzSIP is the implementation of did nt, then?
05:15.41jnfullerI agree
05:20.48jnfullerbut from a sip protocol perspective the term DID is markedroid bullshit
05:21.00florzbecause?
05:21.29jnfullerbecause if you actually read rfc's there is only user@host
05:21.58florz.o()
05:22.15florzBTW, no, there is not, not even close
05:22.38florzbut that's immaterial to this discussion
05:23.51*** join/#asterisk sergey (n=sergey@sergey-home.iks.ru)
05:29.29jnfullerHow is <userinfo>@<host>:<port> immaterial to a discussion that asked how a To: field was to be addressed?
05:29.52*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
05:32.58florzI think the discussion was about DIDs, not about "addressing To headers".
05:33.10jnfullerso what's the actual problem, lanning
05:34.44jnfullerlanning: quick SIP question: To: format is <user>@<server>/<did> correct?
05:35.28lanningall inbound calls are going to "s", I thought it was something in the To: field, but it is in the INVITE field.
05:36.18lanningsomehow I need to get the provider to send me the actual DIDs
05:37.01jnfullerare the pastebins from asterisk debug?
05:37.18lanningya.
05:37.27jnfullercan you get one from tcpdump?
05:37.35lanningcore set verbose 10; sip set debug
05:37.41jnfulleror tshark or snort, whatever
05:38.03lanningya, just a sec
05:40.01jnfullers is the default match for unknown, so if your extension matching for the context doesn't trigger what you are sent it is the default match if defined
05:40.44lanningwell the sip debug showed "INVITE sip:s@10.0.1.254 SIP/2.0" so they are sending me "s", not the DID.
05:40.57jnfullerno, that's triaged by asterisk
05:41.05jnfullers is a default extension
05:41.06lanningoh
05:41.25jnfullerthat's why I said you needed to look at tshark, tcpdump or snort
05:41.29jnfullerwhat os are you on
05:42.11lanningCENTOS
05:42.11jnfullerok so tcpdump should be available
05:42.16lanningI'll have a dump in a sec
05:42.47jnfullerif you cap it and look at it in wireshark you should be able to see what they are actually sending in the invite
05:43.57jnfullerand then you just need to check your context extension matching to make sure their presentations are accounted for
05:46.28lanningit has the "s" extension...
05:46.39jnfullerfrom the provider?
05:46.53lanningya.  it's weird
05:46.57jnfullerok that's fucked
05:47.03jnfullerput in a troubl
05:47.31lanningyup, that's next.
05:47.59jnfullerI assume you're getting the call supposed to be to 4086367703
05:48.24florzuserpart s in the request URI is probably perfectly correct behaviour, anything else probably would be badly broken
05:48.35jnfulleris this peered from another asterisk server?
05:48.43lanningya. main number is 4086367700, but the test DID is 4086367703.
05:49.08florzlanning: have you looked at the To-header?
05:49.16lanningI don't know what they are running.
05:49.47jnfullerYou're looking at tcpdump how?
05:50.19lanningtcpdump -i eth0 -s 1500 -w sip.cap host 192.168.22.212 and udp port 5060
05:50.29jnfullerI mean, you're sure you're looking at the wire and not the asterisk chatter?
05:50.29lanningthen pulled into wireshark.
05:50.55jnfullerbecause you will also catch the internal sip chatter for wireshark
05:51.00lanningpacket came from the provider's IP
05:51.47jnfullerok as long as you are sure it is isolated there is zero reason for an upstream provider to be sending you an invite to s
05:52.05jnfullerunless that is a valid username, which in this case is not the case
05:52.52florzjnfuller: could you please justify that statement with RFC references?
05:53.16jnfullerI already did, florz
05:53.30jnfullercut the shit, I',m trying to be helpful
05:53.34florzjnfuller: where? sorry if I missed it ...
05:54.58jnfullerso anyway if the packet did come from the provider it is malformed
05:55.13jnfullerand you should put in a trouble
05:55.34florzjnfuller: am I understanding you correctly that you are stating that the packet is malformed because the userpart in the request URI is 's'?
05:56.01jnfullerNo, I'm saying that if it'
05:56.10lanningflorz, in this case the "s" is not what I am expecting.
05:56.20florzlanning: because?
05:56.34lanningI am expecting the actual phone number, so I can switch based on it.
05:56.34jnfullerit's supposed to be addressed to 4086367703 and is addressed to s then it is wrong
05:56.55jnfullerso hey this is the tier 1 for asterisk so stop being such an ass
05:57.18florzlanning: why are you expecting it to have 4086367703 as the request URI's userpart?
05:57.21jnfullerlanning is here to get help
05:57.37lanningbecause that is the number that I dialed
05:57.48florzlanning: through the PSTN?
05:57.53lanningyes
05:58.09lanningto the provider, which should be passing it to me
05:58.35florzlanning: well, from that alone you can't expect that - if the provider didn't tell you they would do it, it's rather unlikely that they will
05:58.36lanningcell phone -> PSTN -> ITSP -> Asterisk
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05:58.54kaldemarlanning: do you register to your provider, and if so, how?
05:58.56florzlanning: I assume that you are registering to the provider (via SIP, I mean)?
05:59.07lanningyes
05:59.22lanningvia user.conf
05:59.34lannings/r/rs/
05:59.36florzlanning: well, then you are telling the provider in the REGISTER request which SIP URI to send calls for you to
06:00.14florzlanning: and there, you are probably telling it to send calls to s@<yourmachine>, which the provider, abiding by the standards, does
06:00.19jnfullerdoes the context you register for that user have any sort of pattern match for this number?
06:00.34lanningyes
06:00.44jnfulleris there a wildcard match?
06:00.50kaldemaryou should do it in sip.conf with "register => username:secret@host/callbackextension"
06:00.55lanningexten = _4086367701,1,Goto(default|522|1)
06:00.56lanningexten = _4086367703,1,Goto(default|510|1)
06:00.56lanningexten = _X.,1,Goto(default|500|1)
06:01.04kaldemarif you're lacking the callbackextension, the provider will send s.
06:01.13florzlanning: so, there is nothing else the provider can do in this case
06:01.41florzkaldemar: that doesn't help, as it doesn't allow different number to be distinguished
06:01.55lanningso, I have to register every DID I have?
06:01.58jnfullerthe provider doesn't actually send s, but it sends unscreened
06:02.03lanningI have never needed to do that before
06:02.17jnfullerso unless you have defined uac's asterisk will triage them as unknown
06:02.36florzlanning: either that, or look at headers - which is why I asked you whether you had looked at the To: header ...
06:03.00lanningthe To header has the correct sip uri, the INVITE does not.
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06:03.23florzlanning: the INVITE does have the correct SIP URI, just not the one you are expecting
06:03.43florzlanning: it's the only one allowed there by the standard
06:04.20lanningso if I have 500 DID's I have to have 500 register lines? and register 500 times?
06:04.46florzlanning: no, you have to look at the headers, still
06:04.58florzlanning: or not use registration, if your provider offers that
06:05.18lanninghow do I get asterisk to use the To header in extensions.conf?
06:05.45florzlanning: basically: not
06:06.07florzyou'll have to do that yourself with your dialplan code
06:06.11lanningthen do tell me I have to look at the To header.
06:06.40jnfulleryou could wildcard match all calls and redirect a callerid num to a new context but that has the tendency to be buggy
06:06.42lanningI have never had this issue before, why just this provider?
06:06.51florzwell, I am just telling you what you'll have to do from the point of view of the protocol
06:07.02rue_moreI think I'm getting t the point where I can make my own co interfaces
06:07.22rue_moreanyone want ulaw pots interfaces?
06:07.25florzlanning: maybe the other providers didn't implement SIP?
06:07.49rue_moreheh use a scsi bus, haha
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06:08.06lanningsip register worked just fine, and I got all the DIDs working, straight.
06:08.21lanningjust this one is an issue.
06:08.22florzlanning: well, then you weren't usind SIP
06:08.48lanningwell it was asterisk to a sip provider.  that's all I can say.
06:09.06jnfullerdo you have a peer set up in sip.conf to receive calls unregistered from the provider?
06:09.41florzlanning: well, no, obviously it wasn't a sip provider, had it been a sip provider, it hadn't worked
06:09.50lanningno just the users.conf with insecure = very
06:10.03lanningflorz, go away if you are not helping
06:10.35jnfullerwhat revision of asterisk is this you are using
06:10.37florzlanning: oh, sorry, I am not helping by explaining the functioning of SIP?
06:10.53lanning1.4.18.1
06:11.18jnfullerset up a friend using the provider IP with insecure=port;invite
06:11.29jnfullerer port,invite
06:11.31*** join/#asterisk Winkie (n=urmom@ur.fa.gs)
06:11.48lanningok, just a sec
06:11.52jnfullercreate it in an isolated contect
06:11.55jnfullercontext
06:12.35florzlanning: and anyhow, I guess am capable of deciding when I want to go away all by myself, but thanks for your support ;-)
06:13.39jnfuller[peer]
06:13.39jnfuller<PROTECTED>
06:13.40jnfuller<PROTECTED>
06:13.40jnfuller<PROTECTED>
06:13.40jnfuller<PROTECTED>
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06:14.22jnfullerand then try to match in the captive context while not registered
06:14.34jnfullerunless your provider forces registration
06:15.33florzjnfuller: you aren't seriously trying to tell lanning to not use registrations, are you?
06:15.58jnfullerdepends on the provider
06:16.15florzwell, yeah, of course it depends on the provider
06:16.28florzbut there is nothing to "try out", obviously
06:16.41*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
06:16.46*** join/#asterisk sgupta` (i=3ba0e014@gateway/web/ajax/mibbit.com/x-d37e2547b5c94b2b)
06:17.17sgupta`hello
06:17.28jnfullerthe definition still works if he is registered, right?
06:17.44lanningok, never mind, I will have to call the provider in the morning.
06:17.45sgupta`any indian users here on the channel
06:19.54florzjnfuller: depends on what you mean by "works", and on what you are trying to accomplish, I guess
06:20.46jnfullerare you nine?
06:21.22jnfullerSeven, maybe?
06:22.07florznope, I am just one person
06:22.14jnfullerso anyway insecure=very is deprecated
06:22.30jnfulleras in a seven-year-old, fucktard
06:22.41*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
06:23.14lanningok, I will steer away from "very"
06:23.15florzYou mean, like, I don't use such words as you are using yet, so I must be underdeveloped somehow? I see.
06:23.27jnfulleryou've done nothing but pick apart anything anyone says in trying to help
06:23.32jnfullervery immature
06:23.43*** join/#asterisk oej (n=olle@212.17.152.150)
06:24.58jnfullerif this is the sort of tier 1 asterisk users can expect then no wonder people go with commercial products
06:25.22rue_mohractually
06:25.24florzYou mean, like, it would be more mature to let you do your guessing completely void of any understanding of the underlying technology?
06:25.27rue_mohrif you buy digium hardware
06:25.38rue_mohryou get AWESOME support from digium
06:26.03jnfullera lot of what you've said here is completely incorrect
06:26.13florzLike, instead of telling you that you are totally at the dead end, and even what the solution is?
06:26.17rue_mohrjnfuller, but if your critisizing this channels support I wont aregue, and your right.
06:26.32*** join/#asterisk j_kroon (n=kevinc@dsl-244-12-158.telkomadsl.co.za)
06:26.34jnfullerok smartnick what is the solution
06:26.41florzjnfuller: well, could you point me to one incorrect thing I said?
06:27.14jnfullerwell for one, the s in the invite is complete fallacy
06:27.21j_kroonhi guys, was just wondering about mapping SIP codes to call dispositions, preferably without having to adjust the dialplan.  Specifically 404 codes are still resulting in FAILED, I'd like it to result in NOTFOUND if possible.
06:27.24jnfullerfind me the rfc
06:27.48rue_mohrjnfuller, you have  problem? I dont see it back there
06:28.31rue_mohrjnfuller, there are a few people here who are real sharp, dont get me wrong, but there is a lot of background static
06:28.34jnfullerthis is supposed to be a community of people helping each other, and the discussion list usually is but this is appalling
06:28.50rue_mohrj_kroon, cant help, sorry
06:28.53florzjnfuller: well, let me repeat it once again, for the reading-impaired: he is registering sip:x@<hismachine> as his contact address, so the provider sends the INVITEs there. Next try, please.
06:29.15florzsorry, s/x/s/
06:29.23jnfullerso that's triaged by asterisk and defined by asterisk as a dialogue between the provider
06:29.43jnfullerit's not part of the protocol, it's a defined state by a sip dialogue
06:32.22jnfullerbasically, we've been arguing the same point and you've been nitpicking like an idiot
06:32.38florzjnfuller: you mean, like, "the RFC does not require the provider to actually use the location service that users register to"?
06:33.05*** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net)
06:33.40florzjnfuller: well, maybe that's true, but it's completely pointless, as the whole point of having a location service is to have a defined interface for dynamically re-routing SIP URIs
06:33.56jnfullerexactly, but that doesn't mean the provider sends s if you don't define your location, which is what you alluded to
06:34.15dshaphey would anyone here be willing to help me out with the SendDTMF() application?  If I place an outgoing call from my server to my cell phone, have it wait a few seconds, and then SendDTMF(1234), shouldn't I be able to hear the 4 tones?  I don't hear anything.
06:34.26jnfullerthe provider sends to the default dialogue
06:34.49florzjnfuller: no, if you don't "define your location" (you mean, as in, "don't register"?), the provider should reply to any requests with a 404 or something, of course.
06:34.51jnfullerso we're all capable of being wrong or not 100% on our protocols so back off
06:35.25jnfulleryou can register without providing a callback
06:35.36jnfullerwhich is what causes this issue
06:35.44florzjnfuller: what do you mean by "callback"?
06:36.04*** join/#asterisk lmsteffan (n=laurent@114.69.191.239)
06:36.04florzjnfuller: registration uses so-called "contact URIs", if you mean those?
06:36.15jnfullermember:kaldemar
06:36.15jnfuller:
06:36.15jnfulleryou should do it in sip.conf with "register => username:secret@host/callbackextension"
06:36.25jnfullerwe discussed this earlier
06:36.38jnfullersorry for the bad cut and paste there
06:37.21florzjnfuller: and no, you can't really "register without a contact URI", even though you _can_ IIRC send a REGISTER without a Contact header, which can be used to find out the current registrations for an address
06:37.27jnfulleranyway we've driven off the people who needed help so I'm done, this is not what this channel is for
06:37.49*** join/#asterisk xrmx__ (n=rm@host23-250-dynamic.14-87-r.retail.telecomitalia.it)
06:38.18florzjnfuller: and in this case it was all about asterisk explicitly registering sip:s@<hismachine> as its contact URI ...
06:39.21lanningno it wasn't
06:39.23jnfulleryeah but if you hadn't been such a keener asshole out to prove anyone who was trying to be helpful wrong, publicly ridiculing them in the process we might have all learned something
06:40.02jnfullerand after all this lanning still didn't get any relief
06:40.19jnfullerI'm sure florz can take it from here, lanning. He's the expert
06:40.31lanningasterisk is not registering s@host. it was registering 4086367700@host, and when 4086367703 was dialed I got "s"
06:40.41florzlanning: so, your asterisk sent a REGISTER for a different URI than you got the INVITEs to?
06:40.50lanningyes
06:41.08florzlanning: can you paste the REGISTER message somewhere?
06:41.18jnfullermy apologies to everyone not involved in all this stupid discussion
06:41.25lanningnot now, as I am going to bed.
06:41.35dshaphah
06:41.36*** join/#asterisk pcdog (n=pcdog@adsl-130-33.dsl.init7.net)
06:41.41*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-02cad0d8a85e3462)
06:41.50florzwell, that makes helping rather difficult =:-)
06:42.48florzjnfuller: so, you mean, like, it's more sensible to let people tell complete nonsense that won't actually help anyone, just because they are _trying_ to help?
06:43.10jnfullerperhaps some tact may be in order in the future
06:43.11sgupta`I have problems with hangup and answer detection on trunk line, are there any Indian users here?
06:43.23dshaplet me rephrase my question: is there any reason you guys can think of that audio files could be played over my SIP connection and not DTMF?
06:43.31dshapfrom my * box
06:44.21jnfullerand I'm sorry I called you a fucktard
06:44.32jnfullerbut you were being annoying
06:45.33jnfulleryou mean the actual dtmf tones?
06:46.05jnfulleryou need to check your inband dtmf settings
06:46.20dshapthere's this web-based control panel for my SIP provider
06:46.42dshapunder DTMF mode they have auto, rfc2833, inband, and info
06:46.55dshapi've tried enabling each of them and i still can't hear the tones
06:47.15dshapwhich leads me to believe there is something wrong with my server settings
06:47.21dshapi tried dtfmode=auto in sip.conf
06:47.26dshapas well as dtfmode=rfc2833
06:47.37jnfullerwhat about your client?
06:47.49dshapi'm calling a cell phone
06:47.57jnfullerno, where you call from
06:48.06dshapthat's my asterisk server
06:48.07jnfullerthe sip phone, ata, whatever
06:48.13dshapoh
06:48.16dshapi'm using a .call file
06:48.20dshapthat goes to an extension
06:48.32dshapthe ultimate plan is to have it dial another system (not mine) and interact with a voice menu
06:48.42dshapby sending the appropriate DTMF tones at the right time
06:48.44dshapif that makes sense
06:48.50dshapright now it's not sending any tones
06:48.55jnfullerfor an ivr you want to send inband
06:49.05dshap~ivr
06:49.06infobotsomebody said ivr was Interactive Voice Response
06:49.19dshapokay so how do i set that up
06:49.53dshapjnfuller: like if i was going to have my asterisk server dial my cell phone voicemail provider and type in my password
06:49.57jnfullerdtmf inband on both sides, i think, if you're using a call file
06:50.17florzjnfuller: You know, it's annoying as hell when people who obviously haven't read much of the RFCs involved are citing RFCs as an authority ...
06:50.34dshapby both sides you mean my sip.conf has dtmfmode=inband and the web-based control panel for my SIP carrier has INBAND for dtmf mode?
06:50.39dshapnothing else, right?
06:51.05jnfullerI've actually read so many rfc's I have trouble keeping all the info straight, florz
06:51.45jnfullerbut I do work with an engineer who has never read them and keeps talking about dead cisco drafts as gospel, so I can relate
06:51.57jnfullerplus, I've had about twelve beers
06:52.04florz=:-)
06:52.30jnfullerso forgive me if I'm a bit stumbly
06:52.39Qwelljnfuller: Come back tomorrow then.
06:52.40jnfullerI would assume so, dshap
06:53.02jnfullerI've never tried to outpulse dtmf with a callfile
06:53.12jnfullerbut the ivr will need to hear inband
06:53.21jnfullerotherwise it can't react on the tones
06:53.39jnfullerer, use a call file to call a context that outpulses dtmf
06:53.51jnfullerI know a callfile can't outpulse dtmf ;)
06:54.41*** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-82-2.hag.east.verizon.net)
06:55.14dshapjnfuller: i hear absolutely nothing.  and in my CLI output it says that it is executing SendDTMF(1234) and reports no issues with that
06:56.36dshapcould this have something to do with the audio codec configuration?
06:57.15jnfullercan you talk over the channel when you don't use a callfile?
06:58.07dshapi wrote a dialplan that let me call my server from my cell phone, then my server uses Dial() to connect me to another PSTN number, and i can talk through it
06:58.15dshapand hear the person on the other PSTN line
06:58.25dshapso inbound and outbound audio from talking on the channel work fine
06:58.30dshapas does audio from Playback()
06:58.37jnfullerand can you pass dtmf when not using a callfile?
06:58.57dshapi haven't tried
06:58.59dshapi'll try right now
06:59.16dshapim going to make it so that i call my server, press extension 1, and then my server should execute a SendDTMF
06:59.17dshapis that okay?
06:59.47jnfullerI suppose that would work
07:00.13shido6or RTP being blocked
07:00.16jnfullerbut I was more thinking inband all the way like hairpinning an incoming call
07:00.44dshapnot sure what you mean by that
07:00.53jnfullerso you can send the dtmf from your cell to the pstn and check and make sure the dtmf isn't being turfed
07:01.03dshapohhh
07:01.08dshapi can check that also
07:01.12dshapso like
07:01.19dshapmy asterisk server sets up a bridge between 2 PSTN lines
07:01.28dshapand i press numbers on one of the phones
07:01.29shido6or dtmfmode
07:01.35dshapand see if ic an hear on the other side
07:02.19jnfullerright
07:03.16jnfulleronce you know it actually works, it will be easier to figure out why the call file version doesn't
07:03.16*** join/#asterisk fiddur (n=fiddur@c042.rit.se)
07:03.34dshapok we've got issues
07:04.02dshapi made the changes you suggested earlier
07:04.11dshapdtmfmode=inband and inband on my provider control panel
07:04.21dshapand now i can't even dial extensions
07:04.26dshapmy server isn't receiving DTMF anymore
07:04.40jnfullerok so switch them back and make the change in the extension to inband post-dial
07:04.58dshapk 1 sec
07:04.59shido6inband and ulaw work well , rfc2833 and everything else should work
07:05.50dshapthe issue is definitely the INBAND setting on my provider's web contorl panel
07:06.58shido6if its on a ctrl panel there may be some lag between your update and the propogation across their network
07:07.12jnfullerso establish the call using what actually works for your provider and then try forcing SIPDTMFMode(inband) in the context you use for the callfile
07:08.01jnfullercheck the syntax for that, that is off the top of my head. Corrections, anyone?
07:08.16dshapokay sorry i'm very new to all of this and you guys are a little ahead of me right now
07:08.23dshapi really appreciate the help and i'm going to try everything you say
07:08.37dshapi just verified the SendDTMF is not working even when i use it without a call file
07:08.51dshapnow im gonna check the PSTN --> * --> PSTN thing
07:09.24jnfullerdo you know the difference between inband and rfc2833?
07:09.30jnfullerfunctionally, I mean?
07:10.01dshapok
07:10.04dshapno, i don't
07:10.08dshapbut i just made an important discovery
07:10.14dshapher'es what i did
07:10.19dshapi have 2 PSTN lines, A & B
07:10.27jnfullerinband is in the audio path of the phone conversation as audio
07:10.29dshapi used A to call my asterisk box
07:10.41dshap[ok, gotcha]
07:10.51dshapthen in the extension, i used Dial() to call B
07:11.07dshapwhich bridged the 2 channels
07:11.14dshapso they could talk to each other through my server
07:11.16jnfullerinfo and rfc2833 are signalling methods that use the rtp path to send a representation of the digits and not the actual audio
07:11.54dshap[alright so audio SHOULD work because i can talk on the channel]
07:11.59dshaphere's the thing
07:12.05jnfullerwell there's the gotcha
07:12.12dshapwhen i pressed a button on phone B, i could hear the tone on phone A
07:12.14dshapBUT
07:12.15jnfullerpstn sets have their own dtmf tone generators
07:12.18dshapwhen i pressed a button on phone A
07:12.26dshapall i heard was a little blip/click on phone B
07:12.37jnfullerdigital set?
07:12.38dshapit was being cut off or something
07:12.53dshapone is a cell phone, the other is a landline
07:13.02dshapA = landline, B = cell phone
07:13.05dshapbut listen
07:13.09dshapi tried to make the same call
07:13.12dshapwithotu my asterisk server
07:13.17dshapi just picked up my landline and called my cell phone
07:13.22dshapand then the tones worked in both directions
07:13.31*** join/#asterisk stijnbe (n=stijnbe@router.begen1.office.netnoc.eu)
07:13.32dshapmy server or my provider is fucking it up
07:13.56*** join/#asterisk stijnbe (n=stijnbe@router.begen1.office.netnoc.eu)
07:14.20jnfullercould be, some cells have fixed tone durations
07:15.00*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
07:15.07dshapyou think it's working and the issue is that i'm using a cell phone?
07:15.32jnfullerI think that there are a lot of variables to go wrong
07:15.40dshapok i think i've gotten that out though
07:15.46dshapi called my server with my landline
07:15.52dshapand did SendDTMF
07:15.53dshapheard nothing
07:16.29jnfullerso you call and the context answers, does a SendDTMF(1) and you hear no audio
07:16.48dshapyep
07:16.50dshapthat is correct
07:17.01dshapSendDTMF(1234) actually
07:17.02*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
07:17.03florzjnfuller: no, INFO uses the signalling path, which is exactly what sets it apart from RFC2833
07:17.08jnfullertry adding a SendDTMFMode(inband) before that
07:17.19jnfulleryeah that's right, info isn't in rtp
07:17.23dshapokay so the priority immediately before the SendDTMF
07:17.31jnfulleryep
07:17.35dshaptrying that right now
07:17.44jnfulleranyone know what  the actual syntax is for that
07:18.53dshapjnfuller: [Jun  3 00:18:26] WARNING[16811]: pbx.c:1832 pbx_extension_helper: No application 'SendDTMFMode' for extension (intro_menu, 3, 2)
07:19.10jnfullerI think it's setdtmfmode
07:19.26jnfullerjust a sec
07:19.35dshapi dont see it on the application list on voip-info
07:19.43dshapunless it's a channel variable
07:20.52jnfullerjust gonna log into a server
07:21.43dshapk
07:21.50*** join/#asterisk Milad (n=milad@unaffiliated/slackark)
07:21.52jnfullerSIPDtmfMode(inband|info|rfc2833)
07:22.07dshapokay so SIPDtmfMode(inband) then?
07:22.29Miladis any way to manipulate gotoiftime in other date like jalali etc ... ?
07:22.33dshapand my web control panel thing is set to auto for DTMF mdoe by the wya
07:22.44*** join/#asterisk \void\ (n=void@fer227.internetdsl.tpnet.pl)
07:22.51jnfullerbleorgh,n,Set(SipDTMFMode=inband) ???
07:23.05jnfullerlike I said, I am fuzzy on the syntax for that command
07:23.31dshapaccording to voip-info it's an application and not a variable
07:23.37dshapso it'd be SIPDtmfMode(inband) i believe
07:23.38dshapill try it
07:23.57AlmightyOatmealdoes one really need a fxo/fsx channels if one intends to use wifi/voip phones and softphones?
07:24.37jnfullerhttp://www.the-asterisk-book.com/unstable/applikationen-sipdtmfmode.html
07:24.39dshapjnfuller: IT WORKED
07:24.44jnfulleryeah that's the right syntax
07:24.47jnfullercool
07:24.49dshapi heard the tones!
07:24.53dshapdo you think this will work for call files
07:24.56dshapand everything else?
07:25.04dshapshould i just leave dtmfmode=inband in my sip.conf?
07:25.12dshapor do i always have to set the dtmf mode right before i send dtmf?
07:25.21jnfullerwell didn't that break your other calls?
07:25.36dshapi'm pretty sure what broke the calls was the web control panel setting
07:25.38jnfullerah
07:25.41dshapi dunno ill try
07:25.48dshapwith web=auto and dtmfmode=inband in sip.conf
07:26.23jnfullerYou can probably sort it all out through trial and error but to get to an ivr you definitely will need to switch to inband before outpulsing digits
07:26.51dshapyep
07:26.52jnfullerunless the ivr is sip and understands info or rfc2833
07:26.59dshapdoesn't work with dtmfmode=inband
07:27.09dshapgotcha
07:27.15dshapalright now im gonna try this with my voicemail thing
07:27.24jnfullerjust like a person, if the box can't hear the tones it won't work
07:27.33dshapright
07:27.38dshapand if the box CAN hear the tones
07:27.41dshaphopefully it WILL work :)
07:28.38dshapthat shouldn't mess up anything for that channel after i call it
07:28.39dshapright?
07:28.52dshaplike it shouldn't change the way a caller can interact with my extensions by dialing numbers
07:28.53dshapright?
07:29.21jnfullerwell if you do it in the context for outbound for the function you need the change won't be global
07:29.36dshaptrue
07:29.38dshapalright
07:30.04dshapanother quick question
07:30.16dshapyou know the tone you hear before you're supposed to leave a recording?
07:30.22dshaplike, "Please leave a message after the beep"
07:30.24dshapand then you hear the beep
07:30.26dshapis that DTMF?
07:30.59jnfullerI don't think so, I don't know if there is even a standard for that tone
07:31.25dshapwould it be possible to detect it on a channel with asterisk?
07:32.18jnfullerthere is an answering machine detection app but I don't know if it is production reliable
07:32.55dshapah there's some stuff on google i found
07:32.59dshapill check it out thanks for the tip
07:33.44*** join/#asterisk acehunky (n=acehunky@123.252.144.92)
07:33.58jnfullerno prob, glad you got it working
07:34.09jnfullerthe command is AMD
07:37.32dshaphmm
07:37.50dshapthe D option on the Dial() app doesn't seem to work the same way
07:38.09dshapi tried setting SIPDtmfMode(inband) right before a Dial(SIP/number@trunk,20,D(1234))
07:38.29dshapwhich is supposed to send the DTMF before the channels are bridged
07:38.37dshapany ideas?
07:41.25jnfullerperhaps D doesn't respect the settings in the context
07:43.20jnfulleryou might be able to get around that by forcing your sip.conf to inband and then forcing the setting that works every where you dial out?
07:43.39jnfullerugly hack
07:43.43dshaptruth
07:43.54dshapi might not need D
07:43.56dshapjust thought i'd ask
07:44.15dshapto be honest, i really just want to program my asterisk box to leave me a voicemail without calling my phone right now
07:44.21dshapby calling my AT&T voicemail backdoor number
07:44.39dshapi took out a stopwatch and timed how long the wait should be before it starts playback but it doesn't seem to be working
07:45.51jnfullerhmmm why not send yourself an sms instead?
07:46.24dshaphah well the reason i want to learn how to do this is a long and complicated story
07:46.34jnfullerthe best kind
07:46.40dshapalso i have no idea how to send SMS from my asterisk server
07:46.44dshapim sure i would need a special provider for that
07:46.45dshapwhich i dont have
07:46.57jnfullerhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Sms
07:47.16dshapisn't that expensive?
07:47.30jnfullerdepends on your provider, I get free inbound
07:47.40dshaphow much do they charge u
07:47.43dshapoh
07:47.45dshapfree inbound
07:47.45dshapi c
07:47.48dshaphow much for outbound
07:48.23jnfullerI think my plan is something like 200 messages and then 25 cents after that
07:48.41dshapfor how much $ if you dont mind me asking
07:48.46jnfullerbut the ones I send myself through an sms gateway are to the phone so I don't get charged
07:48.47dshapthat's per month i'm assuming right?
07:49.23*** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex)
07:49.30jnfullerI think the equivalent non-discounted version of what I have is around $25 canadian a month
07:49.37jnfulleri work for a provider
07:49.46dshapwait you're talking about your server sending & receiving SMS
07:49.48dshapnot your mobile phone
07:49.49dshapright?
07:50.03jnfullerno I was talking about my plan
07:50.25dshapif i have an AT&T wireless plan with unlimited text messaging
07:50.30dshapcan i somehow get my asterisk server to send SMS using that?
07:51.19dshapi literally know nothing about SMS
07:51.34jnfullerif you want to send out from the phone you will need to use chan_mobile
07:52.26dshapah i c (just looked that up)
07:52.33dshapyea i think that's an adventure for another day
07:52.34dshaphah
07:53.01dshapis it possible that the default SendDTMF between-digit delay is too small for my AT&T voicemail server to be able to understand?
07:53.15jnfullerI went and looked on our server and I'm actually using jabber im and not sms
07:53.39dshapwhen i call my voicemail backdoor, the first thing i hear is "please enter the mailbox number and press #"
07:54.28jnfullertry increasing the tone duration
07:56.17dshapokay wtf
07:56.25dshapi went back to my extension that uses the Dial() app
07:56.32dshapPSTN --> * --> another PSTN
07:56.38dshapand for the other PSTN i put my voicemail backdoor number
07:56.58dshapand when i do that setup, i can't interact with the IVR through the PSTN phone i'm calling on
07:58.13dshapactually
07:58.16dshapthat problem was present all along
07:58.17*** join/#asterisk war9407 (i=war@liquidswords.org)
07:58.31dshapwith my 2 phones that i have
07:58.51jnfullerhmmm swap the two phones and if the problem moves beat one to death with a hammer
08:00.30dshapi tried each phone
08:00.51dshapcan't interact with a PSTN IVR with my Asterisk server as the middle man
08:01.15*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
08:01.18dshapi have a feeling my call file SendDTMF doesnt work either (checking that now)
08:01.24jnfullerwhat are you using for fxs/fxo
08:01.36dshapim not using those
08:01.40dshapit's just an IP connection
08:01.42dshapon the server
08:01.56jnfullerso this is all voip
08:02.43dshapyes
08:02.56dshapas i thought!
08:03.03dshapeven with SIPDtmfMode(inband)
08:03.09dshapwhen i initiate a call with a call file
08:03.25dshapand hook it to an extension that does SIPDtmfMode(inband) and then SendDTMF()
08:03.27dshapit doesnt work
08:04.02dshap(my server is all VoIP....i'm trying to interact with PSTN lines via my SIP trunk provider)
08:05.02dshapso it comes down to this:  i can call my server from my phone and i can hear the tones it sends back if i do SIPDtmfMode and SendDTMF
08:05.04dshaphowever
08:05.09dshapif my server calls ME
08:05.12dshapthe same commands fail
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08:13.49dshapdrmessano: you there?
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08:48.04WeazelONhey guys, i was wondering if anyone knows, how do i change the vm-intro language on a specific extention
08:48.14WeazelONextension*
08:48.37WeazelONi have 2 kinds of language prompts already in the PBX
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08:53.06WeazelONanyone ?
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08:56.05WeazelON:(
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09:05.37jgooI have *8 working
09:05.55jgoohowever there is a delay, about 1.5 rings before it picks up - can I reduce this delay?
09:06.09jgoos/\\r\\n/,
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11:05.28CRCinAUcan anyone help me with what's going on here?
11:05.29CRCinAUhttp://crc.pastebin.com/d1d2d2d5f
11:05.44CRCinAUI'm not sure why I get a broken pipe :\
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11:19.07AlmightyOatmealgrrr, i'm getting a registration timeout for BroadVoice :(
11:21.02devyllany ideea how can I disable "auth-thankyou" messege which is played after the client leveas a message on a queue mailbox ? .
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11:38.21devyllI guess the only way is to replace that audio file with my audio file ?!
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12:14.25shayanyone here using SIP wants to try giving me a call?
12:14.33shayor is there a sip test that calls you?
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12:31.55xa0zIs it possible to have pfsense, and asterisk running on the same box, without problems?
12:32.41[TK]D-Fenderxa0z: As long as pfsense doesn't mess with the ports * needs.
12:34.07xa0zOkay, I'm going to assume no one has really discussed pfsense, and asterisk?
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12:34.50xa0zI'd like to have a single box for my router and voip, but I don't want to get into a big mess trying to make it happen.
12:34.50ms110hello
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12:36.03Great_Anta_Bakain my cdr records my src field is blank. How do I set that?
12:36.43[TK]D-Fenderxa0z: as long as packets get in & out of * unmolested you'll be fine
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12:39.28xa0zReading on pfsense's site, looks like it would be more difficult to set it up on one box.
12:39.33juanIMPgood morning :D
12:41.02[TK]D-Fenderxa0z: when asking if pfsense can coexist ont he same box with *, we all kinda figured yuo even knew how to use 7 manage it.  this does not instill us with a lot of faith
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12:42.19xa0zOk.
12:43.09[TK]D-Fenderdamn a lot fo caps failure today...
12:46.00xa0zWhat would be a good system to use for asterisk?  Something like a ITX, multiple lan, etc.
12:47.12xa0zI have an HP T5700 1.0Ghz 256mb ram, 256mb flash, 1 ethernet, 4 usb (and a usb ethernet adapter).   Would that be suitable?
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12:53.40ariel_xa0z: don't know how big your needs are but, I have a small 5 person office on a P3, 800mhz, 256mg RAM with a 20 gig hdd.  Which due to vm and some menu's it's setup on CentOS 4.7 with about 3 gig of space used.
12:56.09[TK]D-Fenderxa0z: Depends what you're doing with *
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13:18.18wwalkerwhere can I find a list of what the Reason: codes in an Originate response mean?
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13:25.36wwalkerframe.h appears to be the place
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13:34.42ThatKidKelI've got three phones sitting behind a Adtran NetVanta.  They work fine for a bit then go unreachable..  i'm blaming the router closing off the ports, but my customer swears this router "worked fine at another location"
13:34.47ThatKidKelany suggestions please?
13:35.27[TK]D-FenderThatKidKel: qualify=yes <-
13:35.53ThatKidKel[TK]D-Fender..  Already on
13:36.55[TK]D-FenderThatKidKel: You'd have to share actual configs and SIP debug....
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13:39.04rue_mohrisn't there a keepalive thing in there on some stuff?
13:39.15ThatKidKeli fond the issue
13:39.19ThatKidKelcustomer swore that the firewall was off
13:39.26ThatKidKelfirewall was on, but the sip alg was off
13:39.35rue_mohr:)
13:39.53ThatKidKelthey refused to let me into the firewall..  finally i convinced them
13:39.55ThatKidKeli hate people
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13:42.49ajohnsonhas fired customers over such issues :)
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14:00.24jplankcan anyone comment on Rhino's hardware?
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14:03.28[TK]D-Fenderjplank: channel banks are pretty decent, wouldn't use anything else though
14:03.42jplankreally?
14:04.12jplankI was looking at their Ceros servers, matched with their cards. Not a good route?
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14:05.02jayteehello
14:05.08jayteeI am not dave
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14:08.46ricko73jplank: check in #rhino
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14:11.16[TK]D-Fenderjplank: Their cards seem very 3rd tier.  Wouldn't touch them.
14:11.35[TK]D-Fenderjplank: Might be helpful if you actually stated your needs
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14:12.51beekmorning [TK]D-Fender jaytee ricko73
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14:13.05jayteemornin beek
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14:16.15[TK]D-Fendermornin'
14:16.48jplankfender - we have a lot of compatibility issues with the servers we are getting. most of the hardware is pretty generic (supermicron MB, intel processor, micron ram) but still having issues
14:17.01jplanktheir Ceros boxes seemed perfect to my needs
14:18.58[TK]D-Fenderjplank: So ... what.. the RAM doesn't work with your MB?
14:19.00ariel_jplank: I use there channel banks there good. I have used some of there boards and I have had issues and some without.  There support has been very good in helping getting the boards up.
14:19.40ariel_But this point of biz I only use there channel banks.
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14:21.06jplankfender - seems to be just random compatibility issues with linux. All the equipment is redhat certified, and I'm using CentOS, but I can't get anything higher then 5.1 on there
14:21.36jplankThe manufacturer usually gets the issue resolved, but then I'll order the same config, and have the same issues
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14:21.44jplankits too much of a PITA to deal with
14:22.10jplankI'm wondering if I could just go with the Ceros server and maybe sangoma cards
14:22.21[8none1]jplank: I'm a big sangoma fan
14:22.28jplankmost people in here are
14:22.38jplankI should of gone that route in the first place, been using digium cards
14:22.39[8none1]sangoma+polycom and you can rule the world ;)
14:22.58ariel_jplank: We setup allot of boxes up and have been using http://rackable.com/products/c10001u.aspx?nid=servers_00  These for asterisk and we have been using debian lenny.  Software raid.  They have been working our nicely.  Even with the TE420b boards.
14:23.49[TK]D-Fenderjplank: Dell, IBM, HP.  Get a box you can get SERViCE on.
14:25.09jplankmy boss doesn't want to use dell/IBM/hp ect. We currently use ZTSystems, which is a real company, and their support is good, but the back and forth is killing me
14:26.57DarthPointerariel_: have you had any performance issues w/ the software RAID & *
14:27.53jplankariel_: these servers look nice, thanks
14:27.58wackyplDarthPointer: hello, i ask about button click2dial remember ?
14:28.07ariel_DarthPointer: not yet.  We had some issues with ghosting the SAS drives. But we resolved them by setting the images on one drive then creating the raid aftewards.  The issue is drivers from Intel which are also happening on the HP DL160 we use
14:28.08DarthPointerwakcypl, yes
14:29.19DarthPointergotcha; I've got 3 low end boxes I considered adding software RAID to, but havne't gotten started; thanks for the tip about ghosting; what are you using for your imageing software?
14:29.21wackyplDarthPointer: you said to me about asteridex but
14:29.24[TK]D-Fenderjplank: Poor plan
14:29.38wackyplthis application is not talk with browser
14:29.49ariel_jplank: we have not found any servers from dell, hp, Intel to be 100%.  And in fact most SAS drivers are up to RHEL 5.2 not 5.3 yet.  which also means CentOS 5.2 is the latest you can use.
14:30.39wackyplDarthPointer: I'm looking for something like this http://e0800.pl/clickone27012008/clickone.html
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14:31.03DarthPointerwackypl: what application are you looking to use your click to dial for?  (checking)
14:32.08wackyplclick to dial with  can talk on a browser using a microphone and headphones
14:32.25jplankI wasn't trying to get past 5.2 actually
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14:33.25jplankit was 5.2 that was giving me the issue, I'm positive its something the manufacturer is doing though, I could get two identical systems (in every way) yet one will work with 5.2 one wont
14:33.25DarthPointerwacky: sorry, is that IE only?  Can't check it.  So you want a web based softphone?
14:33.27BaylinkIn PRI debug, it appears that the 'arrows' that point in towards the messages are 'Send' and the ones that point off screen are 'Receive'; is that correct?
14:33.47wackyplDarthPointer: DarthPointer yes only IE
14:33.56wackyplDarthPointer: using java & activex
14:34.16jplankfender - whats a poor plan?
14:34.26phelleranyone have experience with app_meetme?  can anyone think of a reason that, given two entries in extension.conf both leading to meetme with the same options would give different results for the same conference # ?
14:36.16DarthPointerwackypl: I haven't deployed anything like that, but here's a link: http://www.pernau.at/kd/voip/ActXPhone/ to an ActiveX SIP Softphone
14:36.35ariel_that reminds me I need to ask Patrick what's up with the board testing he is doing for us....
14:36.37DarthPointerwackypl: I would start with the list here: http://www.voip-info.org/wiki/view/VOIP+Phones, under soft phones though
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14:37.52wackyplDarthPointer: THX
14:38.05wackyplDarthPointer: I HAVE TO GONE, i write to you later
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14:50.22wwalkeranyone had any significant success in automating a "wait until the callee's voicemail is recoding to leave our message" ?  This is asterisk initiating the call (AGI, or dialplan) and if the other end is an answering machine, detecting when the other end probably just started recording
14:50.58[TK]D-Fenderwwalker: WaitforSilence()
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14:51.05wwalkerAll the searches I do tell me about issues regarding asterisk voicemail, no asterisk calling someone else voicemail.
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14:51.33wwalker[TK]D-Fender: got any idea what an optimum wait time is?
14:51.57[TK]D-Fenderwwalker: "till its done"
14:53.31wwalkerproblem there is some answering machine will hang up because we were silent too long
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14:57.05DarthPointerwwalker: this was jsut discussed last night on list; Answering Machine Detect- http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD
14:57.18DarthPointerwalker: although there was some question as to weather or not it was enterprise grade
14:59.08[TK]D-Fenderwwalker: WaitforSilence() <-------
15:00.29wwalkerDarthPointer: thank you
15:01.50wwalker[TK]D-Fender: uh, yeah, there are 3 variables.  the first one requires the correct setting, which varies for every product that might answer the phone and varies according to whether or not the person records chunks of silence in their greeting.
15:03.08ricko73is there a way when building Asterisk to have it grab the sound files locally instead of downloading them every time?
15:05.35wwalkerDarthPointer: which list?  I don't see anything about answering machine detection in the archives for asterisk-user
15:06.02kaldemarricko73: prevent the download by de-selecting sound packages in make menuselect
15:06.11DarthPointerwwalker; sorry, on IRC last night
15:06.43ricko73kaldemar: I don't think you understand me.  We need the sound files, but want the build process to grab them locally
15:07.39ricko73if they already exist on my computer, why should I have to download them every time I compile a new version
15:08.31kaldemardo you already have the files in the right place before compilation?
15:10.00kaldemarif not, just copy them to the right directory.
15:12.25ricko73kaldemar: what do you mean by "the right place"?  In the target directory?
15:13.02ricko73or is there a location in the asterisk-XXX/ build directory that the downloaded sound.tar.gz files could be placed to prevent the download?
15:14.05[TK]D-Fenderwwalker: AMD first, WaitForSilence() second
15:14.15kaldemarricko73: i mean sounds directory by the right place, /var/lib/asterisk/sounds by default.
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15:15.01spckwhat do you guys think of virtualizing asterisk with virtualbox?
15:16.27kaldemarricko73: you could try to put the sounds tarball into asterisk-x.x.x.x/sounds/ that's where it downloads them.
15:16.27ariel_spck: just one thing comes to mind.... have a nice test lab....
15:16.48ricko73kaldemar: that's closer to what I thought
15:16.51spckariel: but not as an actual deployment?
15:17.02ricko73this is an automated build process for a cross-compiled system
15:17.15ariel_spck: depends on what you are doing with the asterisk
15:17.24ricko73that's why I'm trying to prevent downloading several megabytes every time we re-build
15:17.50spckbasically i want to load balance between two physical boxes for small call center
15:17.59ricko73it just slows things down.  I'll modify our Makefile to copy the sound files into that directory and see if it works
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15:24.51iratikI can think of several ways of doing this -- but i'm just wondering if there isn't something out there that i've missed that does this already. I need to balance the number of calls that are going out between 4 trunks. Usage patterns favor that the first outbound trunk in a dialplan gets the most traffic, then 2nd, then 3rd.  I suppose i could do some sort of cycling on the trunk order based on section of the hour and use that in the dial plan ... and
15:24.51iratik<PROTECTED>
15:26.50iratikeverything i'm thinking of seems awefully hackish though
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15:31.05acxtyHi guys
15:31.19acxtyIs it possible that asterisk automatically makes a call
15:31.33acxtyFor example I want that it call me all the days at 5PM
15:31.37acxtycan that be done?
15:32.04russellbYes.
15:32.13acxtyrussellb, where can I find information on that
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15:33.51russellbAsterisk has multiple ways that you can tell it to originate a call.  You need to write a script that tells Asterisk to originate calls at the right time.
15:34.10russellbYou can do via a TCP socket (AMI / manager interface), or via dropping files into a directory (call files)
15:34.27russellbor the CLI ... but that's not intended for scripting.
15:34.57acxtyrussellb, thanks will search on that ;)
15:35.12russellbyou're welcome.
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15:37.16kaldemarwould be nice to have an optional call time parameter in call files. it would make timed operations much easier.
15:37.31ricko73kaldemar: that did the trick.  Thanks.  Not sure why I didn't think of copying the files there before
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15:42.36Zhadkal> don't you do it by setting the timestamp of the .call file?
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15:49.22[TK]D-FenderZhad: you do
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16:03.56kannanhello all, i am using phpagi for an AMI script to originate. All is ok, excet that I have an AGI as Application and myagi|followed by ARGS in DATA in the ORIGINATE. The DATA arguments is getting truncated, is this unavoidable?
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16:04.38Kernel_Corehi all
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16:06.04*** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi)
16:06.58Kernel_CoreI am running latest version of libpri 1.4.10 and dahdi 2.1.0.4  , I have 2 E1 links which are connected to TE110P , I configured both E1 , when I send traffic to Second E1 , it is OKey ! but when I send traffic to the First E1 link , it works but after 1-2 hours it doesn't response and I have to run dahdi_tool and LOOP it ! what can be wrong ?
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16:12.48[TK]D-Fenderkannan: How long?
16:12.52Kernel_Coreand here is my pastebin for debug E1
16:12.53Kernel_Corehttp://pastebin.ca/index.php
16:13.00rue_mohrso, paging, I think how that works is that there is a sip account on the phone thats set to auto pickup to speakerphone in a huge group call with one way audio?
16:13.04[TK]D-Fenderkannan: Don't forget the base dialplan parameters have a limt for sure themselves
16:13.25[TK]D-Fenderrue_mohr: Generally, no.
16:14.25rue_mohrarg, so, in general how does it go?
16:15.15[TK]D-Fenderrue_mohr: A SIP header is sent with the call to tell it to auto-answer on speaker.  the end.
16:15.38[TK]D-Fenderrue_mohr: *'s Page() is just a multi-callout + Meetme w/ 1-way audio
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16:24.10rue_mohroh, so most of the hard stuff is already done
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16:26.03timeshell_atworkHappy Threesday!\
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16:29.40ctooleytimeshell_atwork, is that something like Three-Times-As-Much-Stuff-Will-Break-Today-Day?
16:30.09timeshell_atworkctooley :  No.... Threesday comes after Twosday?
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16:33.40enzohi
16:33.50rue_mohrbrb
16:34.14enzoI'm doing an upgrade of my asterisk, but it crashes whenever i receive a fax, in faxt RxFax crashes asterisk, any idea how to solve this problem ?
16:34.21mort_gibctooley: I'll buy that explanation !!
16:34.27enzoi use asterisk-app-fax
16:43.13coppiceenzo: you probably have multiple versions of spandsp on your machine
16:43.28enzono only libspandsp1
16:43.48enzoit's a fresh install of ubuntu jaunty in fact coppice
16:44.26coppiceubuntu may have installed a version of spandsp. I assume you installed a more recent one
16:45.00enzono I haven't upgrade to a new libspan version
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16:45.51enzoi have libspandsp1  0.0.5-pre4-1 on y ubuntu (default version for ubuntu 9.04)
16:46.30coppicethat's pretty ancient, but I think * is supposed to allow for that version OK
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16:47.41enzowell the asterisk-app-fax is version 2007-06-24, quite old also
16:47.49kannan[TK]D-fender, i am not able to find the docs that specify the limitastions for the base dialplans themselves
16:47.57enzodon't know why a fresh ubuntu gives so old packets
16:48.14tzafrir_laptop(trivia: libspandsp.so.3 of Debian packages is actually older than libspandsp.so.1 which is the same as upstream's SONAME)
16:48.19kannan[TK]D-fender, i am sure thats the case, as the values are very lengthy, sometimes whole sentences
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16:48.45kannan[TK]D-fender, i guess i will do it with DB tables,
16:48.48tzafrir_laptopenzo, and won't build with that specific version of spandsp, IIRC
16:48.54rue_mohrso now Ihave to see if voip-info has a paging page
16:49.14tzafrir_laptopA known and painful issue of the sid freeze. Just use app_fax :-(
16:49.19enzotzafrir_laptop: maybe you have an idea for the fact that rxfax crashes ?
16:49.39jeffspeffi have a 4 "line" ip phone. what is the recommended way to have the "lines" work like regular pstn lines for answering? example: call comes in, line 1 rings and anybody can answer line 1; another call comes in, line 2 rings and anybody can click "line 2" on their phone and answer.
16:49.39kannan[TK]D-fender, thansk a lot, ; can you kindly point me to some resources that explain the base application limitations, i am searching it in voip-info.org
16:49.40tzafrir_laptopbecause it was built for a different version of spandsp?
16:50.00rue_mohrjeffspeff, you want it to act like a keyed system dont you
16:50.16jeffspeffif that's the term, then yes
16:50.23DarthPointerkernel_core; you did not paste the correct url
16:50.24rue_mohrjeffspeff, asterisk isn't a keyd system
16:50.50rue_mohrjeffspeff, DONT use polycom sets... that will save you a lot of hurt
16:51.26rue_mohrjeffspeff, there is something called sip2, which can try to emulate a keyed system, your luck may be limited
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16:51.44jeffspeffrue_mohr, i thought that * would have been able to do that
16:52.22enzothat's strange ubuntu made such an error tzafrir_laptop, it should have already been reported, and i find very few things on this
16:52.23laskoIs there a difference in how RealTime works in 1.6 as compared to 1.2, 1.4? I seem to be having trouble with getting it setup with 1.6.
16:53.04jeffspeffrue_mohr, what if i were to create a "line 1 user", "line 2 user", etc. and have that sip user registered on the respective "line" on each phone. would that work?
16:53.41tzafrir_laptopenzo, I'm not really sure packages in Universe get QA
16:54.29enzois there a way for me to know what version of wich soft I have to install tzafrir_laptop ?
16:55.41enzoin fact when it crashes, i can see this error tzafrir_laptop : asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_set_local_ident
16:56.26rue_mohrjeffspeff, I just been thu this, I can tell you what I did, and all hte problems with it
16:57.25[TK]D-Fenderkannan: length limits hit people doing big pages, etc, and I recall one guy submitting a patch to extend it a LOT, and it is needed.
16:57.44[TK]D-Fenderkannan: I don't know the hard number myself, but I'm sure you could generate an answer really easily
16:57.51rue_mohrjeffspeff, they will tell you, and from hindsght I will too, try to start by ditching everything you know about phone systems in keyed context, start with the notion that every phone call has a specific owner, and that for another user to have that call, the origional user must relinquish it
16:58.26rue_mohrjeffspeff, if you start with that, then you can have a system that isn't much different than your trying for, but isn't a keyed system
16:58.50rue_mohrjeffspeff, I know its not what you dont want to hear, but did you?
16:58.53[TK]D-Fenderjeffspeff: What phone are you looking at?
16:59.08jeffspeffjust one sec. on the phone with customer
16:59.13rue_mohrjeffspeff, nomatter how much tk tells you to get polycom phones, DONT DO IT
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17:01.21xorlHey guys, what's a good sip provider besides vitelity the service is anoying.
17:01.22[TK]D-Fenderjeffspeff: Nothing rue_mohr gets his hands on works, and refuses to retun patently defective goods insisting "I MUST be able to get them to work".  It aint the tools, its the SMITH.
17:02.13jeffspeff[TK]D-Fender, i was looking at the 4 line polycom phone
17:02.52rue_mohra) its not a 4 line phone, it just has 4 line keys
17:03.02rue_mohrall polycom phones have too few buttons
17:03.08rue_mohrand they have severe dtmf issues
17:03.21rue_mohrand the documentationa and support sucks
17:03.29[TK]D-Fenderrue_mohr: Only for you really.  No-one else in here seems to share your issues
17:03.29rue_mohrwhat am I up to D)?
17:03.49rue_mohr[TK]D-Fender, you didn't do that goodle search for polycom dtmf asterisk
17:03.51rue_mohrI did
17:03.53jeffspeff[TK]D-Fender, do you think that the idea of registering users for the lines, and have the call come in to line 1 (user 1), if line 1 is busy, go to line 2 (user 2) instead of vm, then line 3, etc.
17:03.54[TK]D-Fenderjeffspeff: What model?
17:03.59rue_mohreveryone has a problem with polycom dtmf
17:04.35jeffspeff[TK]D-Fender, soundpoint 440
17:04.38jeffspeff*550
17:04.39Qwellrue_mohr: name one other person
17:04.45[TK]D-Fenderrue_mohr: really?  Where are all these screaming frustrated users?  not HERe, and you know they are the most advocated maker here.
17:04.57rue_mohrI dont want to post the 160000 google hits
17:05.08rue_mohrno, 1% of techs use irc
17:05.15[TK]D-Fenderjeffspeff: IP 550 is very hard to recommend in their line-up.  What kind of use will the actual user make of it?
17:05.39jeffspeff[TK]D-Fender, i think i read somewhere that you can register a seperate user for each "line"
17:05.45rue_mohrpolycom phones also take up too much deskspace, and where the hell are my wall mount brackets!?
17:06.05ariel_Loves and uses mainly Polycom phones.. And has no dtmf issues with them....
17:06.10rue_mohrjeffspeff, in a month, I'll still help you get it working the way your asking for
17:06.13[TK]D-Fenderrue_mohr: When you have to say "go look", that tells us this "problem" hasn't made it to the people who run large * installs here.
17:06.21Qwellrue_mohr: http://www.888voipstore.com/polycom-wall-mount-kit-pr-18556.html
17:06.34rue_mohrI ordered some, its been 3 months
17:06.45rue_mohrfrom williams communication
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17:06.58[TK]D-Fenderjeffspeff: You looking to share a single phone amongst multiple users?
17:07.12rue_mohrhah no
17:07.14[TK]D-Fenderrue_mohr: WILLIAMS is your problem then
17:07.19rue_mohrhe's trying to emulate a keyed system
17:07.27[TK]D-Fenderrue_mohr: Not yet.
17:07.33jeffspeff[TK]D-Fender, the 501 would work just as well. i just liked the 4 "line" part.... it's for a doctors office. each persone will have there own phone.
17:07.56[TK]D-Fenderjeffspeff: Ok, then throw the entire concept of "lines" out the door.
17:07.59rue_mohryour "4" line phone can take 12+ lines
17:08.18rue_mohrI'm sorry, I'll shut up, for now
17:08.18jeffspeff[TK]D-Fender, i understand that sip uses channels etc.
17:08.30[TK]D-Fenderjeffspeff: the term "lines" by these makers implies the number of unique IDENTITIES your phone can have.  In your case your phone is for just 1 person
17:08.36rue_mohrsits on the sidebench and watches this all unfold again
17:08.45[TK]D-Fenderjeffspeff: thus its jsut down to how many CALLS you can juggle at a time.
17:09.39[TK]D-Fenderrue_mohr: the start of his question wasn't a demand for key-style access, just thinking in those terms before learning any different
17:09.55[TK]D-Fenderrue_mohr: We'll see the reaction one the new reality settles in.
17:10.00rue_mohrI bet he's gonna get a tdm400 card and have terrible echo problems
17:10.35[TK]D-Fenderrue_mohr: Any more predictions to share with us Nostradumbass? ;)
17:10.39rue_mohrbut I have to say, digiums support is AWESOME
17:10.46kannan[TK]D-fender, thanks again, i have still not got the actual limitations number, i will update when i do,
17:10.52jeffspeff[TK]D-Fender, ok so lets say call comes in and is routed to person A. But, person A is making some copies right then, how would person B pick up that call? They're wanting to do it in the pstn way of person B clicking the "line 1" button on phone and answering
17:10.53[TK]D-Fenderrue_mohr: Yes, you are having to work with them daily ;)
17:11.20[TK]D-Fenderjeffspeff: When you call person A and they aren't there, isn't that where Vm kicks in?
17:12.03jeffspeff[TK]D-Fender, yes, usually, but they don't want it to go to vm right away during business hours if somebody else is able to answer the call.
17:12.36rue_mohrjeffspeff, may I ask what kinda phone system they use now?
17:12.50[TK]D-Fenderjeffspeff: * has  call-pickup capabilities, but it isn't so simple as grabbing a "line button"
17:12.58jeffspeffrue_mohr, some shitty pstn
17:13.16[TK]D-Fenderjeffspeff>rue_mohr, some shitty pstn <- huh?
17:13.17mort_gibI just upgraded asterisk from 1.4.24 to 1.4.25
17:13.38mort_giband I get loader.c: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: ast_smdi_interface_find when reloading, and voicemail is not loading??
17:13.51jeffspeff[TK]D-Fender, i don't know what kind of system they have now, except that they don't like it, it's proprietary, and uses pstn lines
17:13.58[TK]D-Fendermort_gib: Plenty of nasty bugs in that rel, you may want to downgrade till .26
17:14.07mort_gib:-(
17:14.23mort_gibHaving probs with this install....
17:15.15mort_gibDowngrade to 1.4.26 ?? BUt that's in rc is it not??
17:15.40rue_mohrjeffspeff, do the phones say something like 'nortel' or 'meridian' on them?
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17:16.19[TK]D-Fendermort_gib: ... downgrade to 1.4.24 UNTIL .26 gets released
17:16.21rue_mohr'panasonic' ?
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17:16.38[TK]D-Fenderjeffspeff: Ok, that could be ANYTHING.
17:16.48mort_gib[TK]D-Fender: Ok, thanks!
17:16.52jeffspeff[TK]D-Fender, my idea was to use the 550 which has 4 lines. make sip users for "lines" 1,2,3. have the actual user registered to "line button 4". set up dialplan so that incomming calls go to user "line 1" which will be a registered user on the "line 1" button on all phones. that way anybody can press the "line 1" button and pick up the call. if "line 1" user is busy, then call roll over to "line 2" user instead of vm.
17:17.22rue_mohrholds his mouth
17:17.24jeffspeffand "line 2" user would roll to "line 3" user in same way.
17:17.40jeffspeffor am i way off in how i'm thinking about this?
17:18.00[TK]D-Fenderjeffspeff: How many phones, how many lines?
17:18.01rue_mohr[TK]D-Fender, yea hows he doin?
17:18.15[TK]D-Fenderrue_mohr: Still a little jumbled :)
17:19.26jeffspeff[TK]D-Fender, about 8 phones. # of lines doesn't matter because it's all voip/sip
17:20.38[TK]D-Fenderjeffspeff: there are some ways to emulate this.  Just for pickup you don't need anything configured on the phone itself you just dial a special ext # and it will grab the call.
17:21.11[TK]D-Fenderjeffspeff: But forget the concept of "lines".  There is no association of "line" toa  call going to a phone.
17:21.26jeffspeff[TK]D-Fender, can you direct me to a link on how an ext like that would work?
17:21.36jeffspeffor the name of the feature?
17:21.45[TK]D-Fenderjeffspeff: Go lookup "call pickup" on the WIKI
17:21.47[TK]D-Fender~wikis
17:21.47infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
17:22.12jeffspeff[TK]D-Fender, ok, thanks for your help
17:22.58rue_mohrI dont think he really hear all that stuff I said
17:23.39Qwellrue_mohr: You know that guy on the street with the big sign shouting about the apocalypse?  You know how people just walk by and ignore it?
17:24.33rue_mohrI'm interested to know if he takes the same path I did
17:25.02[TK]D-Fenderrue_mohr: And I saw you prjecting your needs before he gained a better understanding of the natural call handling style of these phones with *
17:25.28rue_mohrto me, right off the bat he said he wanted a keyed system out of asterisk
17:25.44rue_mohrand he had looked some stuff up to know to say and ask what he did
17:26.20jeffspeffrue_mohr, i said if that was the term for it... and i didn't look anything up, i'm not completely new to * or sip
17:26.20[TK]D-Fenderrue_mohr: Your "wants" became inflexible requirements and the entire process was a fight and you still continue to face the proper corrections your situation calls for.
17:26.48[TK]D-Fenderrue_mohr: So keep on pounding that square peg into the round hole :)
17:27.09[TK]D-FenderQwell: What guy?
17:27.14rue_mohrand I made it fit!
17:27.27rue_mohrthe peg and the hole both took a bit of a hit in the process
17:27.38[TK]D-Fenderrue_mohr: congratulations.... how's that card of yours doing?
17:27.53xorlhmm, so I need a new DID provider, I have vitelity right now but I am constantly having quality issues with their service, anyone recommend me something similar?
17:27.57rue_mohrwell, since I called digium support and talked to them its a lot better
17:28.26[TK]D-Fenderrue_mohr: How "passable" is it now?
17:29.20rob0Hmmm, vitelity is working here. What quality issues?
17:29.39rob0Oh, for DID I just use ipkall :)
17:29.56rue_mohron friday I'm gonna apply a patch that should resolve most of the audio issues
17:30.25xorlrob0: well I constantly get quality issues, echo etc.
17:30.30*** join/#asterisk Khratos (n=khratos@190.166.103.254)
17:30.36*** join/#asterisk BlackSlik (n=james@41.219.218.217)
17:31.12*** join/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br)
17:31.33Nilzaohi guys
17:32.31rob0echo, couldn't that be an issue with the phone itself?
17:32.41*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:32.55*** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net)
17:33.02NilzaoI can't find if * is able to host E1 port
17:33.07Nilzaois that possible?
17:33.12[TK]D-Fenderrue_mohr: "should".  So not yet. How far off are things?
17:33.28[TK]D-FenderNilzao: It can't.  Asterisk is SOFTWARE
17:33.49[TK]D-FenderNilzao: However Asterisk supports all sorts of HARDWARE that will let you interface with an E1
17:34.08Nilzao[TK]D-Fender: ok then i can tell the interface E1, that i'm hosting lines?
17:34.23jeffspeffxorl, have you looked at Teliax?  http://teliax.com
17:34.29[TK]D-FenderNilzao: what does "hostling lines" mean?
17:34.42rue_mohrwell, some of the audio is a little funny, BUT they resolved the muting issue that none of you could help with
17:34.52rue_mohrseems that there isa setting in the echo canceler
17:35.22Nilzao[TK]D-Fender: my first time with E1 ports, i use FXO and FXS... to me FXO can access analog line, and FXS "hosts" an analog line
17:36.04rue_mohr<PROTECTED>
17:36.06[TK]D-FenderNilzao: E1 is a digital trunk.  * can talk to it directly witha  wide variety of ahrdware.  what do you want to DO once * gets the call?
17:36.24rue_mohrits usually set to 24, 3 or 4 works great to stop it from muting during the conversation
17:37.03[TK]D-Fenderrue_mohr: Gues no-one has your issues with the card, and evidently this is still a work in progress with the tech of the maker themselves.  Guess you can't expect the community to beat that, and those techs themselves have not finished the job even working directly with you
17:37.14Nilzao[TK]D-Fender: have 2 E1 ports on * , and 1 E1 PBX
17:37.41Nilzao[TK]D-Fender: the first E1 port on * receive outside calls
17:37.44[TK]D-FenderNilzao: Ok, so are you looking for * to sit BETWEEN the telco & your other PBX?
17:37.59*** join/#asterisk mykhyggz (n=mykhyggz@evolone.org)
17:37.59[TK]D-FenderNilzao: Ok, sounds like "yes"
17:38.05Nilzao[TK]D-Fender: yes
17:38.15[TK]D-FenderNilzao: so * can act as CPE to the telco, and NET to your PBX
17:38.17Nilzao[TK]D-Fender: just say if is that possible
17:38.19*** join/#asterisk s0lid (n=s0lid@122.53.104.57)
17:38.25[TK]D-FenderNilzao: Of course its possible
17:38.47Nilzao[TK]D-Fender: thanks, any keyword to google it?
17:38.54*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:39.10[TK]D-FenderNilzao: What is there to goole?
17:39.12[TK]D-FenderGoogle*
17:39.18ricko73I wonder if we had a discussion about green jello vs red jello if the results would be different than "what's a good IP phone"
17:39.27Nilzao[TK]D-Fender: I can't thinkn how i ask it to google...
17:39.37Nilzao[TK]D-Fender:  please google, how i host E1 ports
17:39.41[TK]D-FenderNilzao: ask WHAT?
17:39.47rob0um ... should we party with lemon jello?
17:39.48Nilzao[TK]D-Fender: or kinda something
17:40.16[TK]D-FenderNilzao: You don't HOST.  this is not a magic word!  You are either CPE or NET.  go look at the Zaptel/DAHDI setting for "signalling" and jsut pick the right one.
17:40.16Nilzao[TK]D-Fender: i will try your way "* between telco and PBX"
17:40.53[TK]D-FenderNilzao: Drop the word "host".  It does not apply to this application at all.
17:40.54Nilzao[TK]D-Fender: now i know what to ask for google... CPE or NET tx
17:41.42dshap[TK]D-Fender: I've got a question but I don't want to interrupt...would now be okay or wait till you are done talking with Nilzao?
17:41.55[TK]D-FenderNilzao: And I jsut gave you the parameter that is different between setting one up VS the other.  Also your span (again working on the premise that you are using a Zaptel-type card) should take timing from the telco side, and set it for the PBX side
17:42.00Nilzaodshap: i'm done here thanks
17:42.14[TK]D-Fenderdshap: Just ask out loud unless you have special reason to ask me directly
17:42.28[TK]D-Fenderdshap: And Just ask if you're going to ask.
17:42.33dshapokay well you're the one who always knows what's up around here hah
17:42.34dshapbut yeah
17:42.44dshapi'm having issues passing DTMF tones through my SIP trunk it seems
17:42.46dshapwith my Asterisk server
17:43.02dshapbasically i'm trying to get my Asterisk server to dial an outgoing call and interact with an IVR through SendDTMFs
17:43.37dshapwhen it didn't work, i thought i'd test the system but calling my asterisk server from my PSTN phone, using an extension that Dial()'s the PSTN IVR, and then trying to control it from my phone
17:43.43[TK]D-Fenderdshap: And yesterday I asked you to show us how you were trying to do it
17:43.52*** join/#asterisk mchou (n=mchou@unaffiliated/mchou)
17:44.01dshapokay since yesterday i've narrowed down the problem quite a bit
17:44.07Nilzao~ivr
17:44.08infobotsomebody said ivr was Interactive Voice Response
17:44.21dshapIf I do PSTN Phone --> * --> IVR
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17:44.31dshapI can't dial DTMF tones on my phone that reach the IVR
17:44.40dshapthey're obviously reaching my Asterisk server since I'm able to interact with my dialplan
17:44.42Nilzaodshap: what phone you using?
17:44.51Nilzaodshap: softphone? ata?
17:44.52dshapi've tried a cell phone and a landline
17:44.55dshapPSTN phone ^^
17:45.13Nilzaodshap: how your PSTN phone access the *? FXS?
17:45.17dshapobviously when i call the IVR directly with my cell phone or landline without putting * into the equation I can interact with it no problem
17:45.28dshapit uses a SIP trunk
17:45.38dshapi call a DID which is tied to a SIP origination provider
17:45.41dshapmy asterisk server is pure VoIP
17:46.01Nilzaodshap: have you tryed relaxdtmf?
17:46.11dshapnever heard of that
17:46.15dshap~relaxdtmf
17:46.22dshaplol neither is the info guy
17:46.27Nilzaodshap: its on sip.conf let me see the right name
17:46.32BlackSlikhow do i make my first IVR with my newly install asterisk box
17:46.42dshapBlackSlik: lol what a question
17:46.50dshapBlackSlik: read the asterisk eBook, that's what i did
17:46.58BlackSlikwhich of it
17:47.04dshap"which of it" ?
17:47.06ariel_BlackSlik: see the sample files there is a demo there
17:47.32Nilzaodshap: try in your sip: relaxdtmf = yes
17:47.50Nilzaodshap: and dtmfmode = rfc2833
17:47.51dshapokay and that's under my specific trunk and not general
17:47.52dshapright?
17:48.02dshapi've tried dtmfmode=rfc2833 before but never with relaxdtmf=yes
17:48.06dshapi should put both in?
17:48.15Nilzaodshap: free to try... just a guess
17:48.20[TK]D-Fenderdshap: No
17:48.21Nilzaodshap: i don't know if it will work
17:48.27[TK]D-Fenderdshap: What phone are you using?
17:48.30ariel_relaxdtmf=yes is for zap
17:48.36[TK]D-Fenderariel_: Not only.
17:48.37dshapcell phone
17:48.38BlackSlikhow do i make my first IVR with my newly install asterisk box
17:48.52[TK]D-Fenderdshap: .....
17:48.59DarthPointerBlackSilk: if you are asking that question you may want to look into some managment software like FreePBX (not supported here, try #FreePBX)
17:49.01dshapcell phone --> SIP trunk --> VoIP Asterisk box --> SIP trunk --> PSTN IVR
17:49.25[TK]D-Fenderdshap: so the lack of DTMF is jsut between your provider and *?
17:49.32DarthPointerBlackSilk: it can kind of pointy-clicky the whole thing up for you :)
17:49.44dshap[TK]D-Fender: seems like the other way around, but yes
17:49.48[TK]D-Fenderdshap: which of those 2 providers has the issue?
17:50.18dshap[TK]D-Fender: i haven't even tried it on Flowroute because I can't even get Playback() to work on Flowroute yet (will get to this later), I'm just using voip.ms at this point
17:50.34dshapmaybe i should try it just for the hell of it
17:50.46[TK]D-Fenderdshap: Do NOT test this with jsut a cell phone, that could be part of the problem.
17:51.02dshapokay well maybe my logic is flawed
17:51.02[TK]D-Fenderdshap: Isolate your tests with voip.ms and use a ahrd landline phone
17:51.05dshapbut i think that's out of the equation
17:51.06[TK]D-Fendersdhit is
17:51.16dshapif i call the IVR directly with my cell phone
17:51.18dshapit works
17:51.24[TK]D-Fenderdshap: WTF IS DIRECT?
17:51.31dshapit has a phone number
17:51.32dshapi call it
17:51.35dshapand i can interact with it
17:51.38[TK]D-Fenderdshap: via what?
17:51.47dshaptouch tones
17:51.52[TK]D-Fenderdshap: Asterisk is software and does nto pickup &#$ing MICROWAVES
17:51.54dshap"Press 3 for Technical Support"
17:51.58dshapi press 3
17:52.04dshapthen it says "Transferring now"
17:52.19dshapif i do it through my * box, it just hangs and then says "Sorry we didn't hear anything"
17:52.25dshap"Please make your selection"
17:52.36[TK]D-Fenderdshap: meantion the exact interface/ service etc these calls LAND ON.  Your description is turning to SHIT so we can't tell what is at faul <---
17:52.45dshapokay sorry
17:52.46[TK]D-Fenderfault*
17:52.48dshapi'll be very upfront
17:52.53dshapi'm calling an AT&T voicemail backdoor number
17:53.09dshapif you call the number, the first thing the IVR asks you for is a 10-digit phone number
17:53.13dshapso you dial the phone number and press pound
17:53.21dshapthis connects you with a user's voicemail so you can leave a message
17:53.37dshapmy ultimate goal is to get Asterisk to leave a message on my voicemail
17:53.41DarthPointerBlackSilk: or for asterisk tutorial see Leif Madsen excellent book: http://astbook.asteriskdocs.org/ http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-15-SECT-3.html#asterisk-CHP-15-SECT-3.6.2
17:53.52dshap(without calling my phone)
17:54.29dshapwhen i call the backdoor number with any phone (cell phone, land line), i can dial in someone's 10-digit phone numebr and it will connect me
17:54.37dshapwhen i call THROUGH my * box (using Dial())
17:54.39xorljeffspeff: you recommend teliax
17:54.46dshapit doesn't receive my input
17:54.51Nilzaodshap: can you see the DTMF in your CLI?
17:54.55dshapyes
17:54.58dshapwell
17:54.58dshapi mean
17:55.02dshapnot as i press them on the phone
17:55.04leifmadsendshap: you probably need to delay it
17:55.15leifmadsenlogger.conf <-- enable DTMF logging
17:55.16dshaplefimadsen: i'm not following...
17:55.22dshapokay
17:55.24leifmadsendshap: what is the Dial() line?
17:55.36*** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-128.oc.oc.cox.net)
17:55.39dshapDial(SIP/8058951743@voipms)
17:55.57leifmadsenright... and how are you then having asterisk dial DTMF?
17:56.12leifmadsenor do you mean you can't even dial dtmf through asterisk manually?
17:56.15Shaun2222anybody seen a weird issue, not sure if it's polycom related but when another calls comming in while your on the phone it mutes out the other person so you cannot hear them, i'm assuming they cannot hear me either.
17:56.18dshapthe latter
17:56.27leifmadsendshap: your dtmf configuration is probably wrong then
17:56.39leifmadsenuse dtmf logging to see if you're getting and sending DTMF correctly
17:56.41Shaun2222it's getting annoying, just started for me, havnt changed anything
17:56.46[TK]D-FenderShaun2222: there is a single momentary cut for the "beep" of CW, but that should be it
17:56.52leifmadsenprobably set to inband when it should be out-of-band (or vice-versa)
17:57.07dshapleifmadsen: so if i enable DTMF logging then I should see DTMF info appear on my CLI as I press them on my phone that's connect to my asterisk server?
17:57.08Shaun2222running 1.6.0.6
17:57.16leifmadsendshap: yes
17:57.20Shaun2222[TK]D-Fender: is that a phone feature or *?
17:57.49[TK]D-FenderShaun2222: Purely the phone
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17:57.58dshaplefimadsen: okay, i will ty this.  but since i can interact with my * server's dialplan/IVR, doesn't that mean it can receive DTMF no problem?
17:58.04Shaun2222actually i did upgrade my phones firmware not long ago, others who have been running on older version say it doesnt seam as long as mine, so that could be the momentary mute for htem
17:58.05dshapleifmadsen: shit i spelled your name wrong
17:58.25dshapleifmadsen: i think the only issue is when it tries to send them back out
17:58.41leifmadsendshap: no worries, everyone says it wrong too
17:58.55leifmadsendshap: dtmf configuration must be wrong then
17:59.02leifmadsendtmfmode=rfc2833 most likely though
17:59.33dshapi've got this "Cutomer Portal" web based control panel for my SIP provider
17:59.36dshapand they have a DTMF Mode setting
17:59.39dshapand right now it's on auto
17:59.46dshapbut i could change it to RFC2833
18:00.01Shaun2222[TK]D-Fender: one other q, is there a way for asterisk to tell the phone to use a different ring, for example i would like calls from the queue to ring one tone but calls direct to there extension to ring a different tone.
18:00.30[TK]D-FenderShaun2222: Yes.  The same way you set the header for auto-answer for paging.  Links on the WIKI
18:00.47Shaun2222thanks
18:01.00dshapleifmadsen: where in logger.conf do i enable DTMF logging?  is it just uncommenting "dtmf" under "[logfiles]" ?
18:01.18leifmadsenconsole => warning,error,notice,dtmf   <--
18:03.26dshapleifmadsen: i enabled DTMF logging.  before Dial() is called (i.e. when i'm just interacting with my own IVR), i see the DTMF logs come up on the CLI
18:03.38dshapafter Dial() is called and the channels are bridged, the DTMF doesn't show up when i press on my phone
18:03.58leifmadsendshap: what version of asterisk?
18:04.21dshapleifmadsen: 1.4.22
18:04.33leifmadsentry latest 1.4 branch to see if you have the same issue
18:04.46leifmadsensome dtmf changes have gone in recently
18:05.01leifmadsenfeel free to check 'svn log' to search for them
18:05.16*** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com)
18:05.32dshapim sorry, im not sure what you're suggesting i do
18:05.37dshap1.4 branch?
18:05.59[TK]D-Fenderdshap: And please learn to test each link SEPARATELY.  You keep combining all this together along with your CELL PHONE.  Prove each little piece ALONE
18:06.07carrarleifmadsen, could you please log into his box and do it for him :)
18:06.13leifmadsencarrar: nope
18:06.16leifmadsen<-- consultant
18:06.18carrarheh
18:06.31neurosys<-- n00b
18:06.34carrar<-- about to take a tour through handford Nuclear site
18:06.34dshap[TK]D-Fender: so far i've not seen any difference at all between my cell phone and my land line, i've tried them both on many of my tests
18:06.42carrar(sp)
18:06.46[TK]D-Fenderdshap: You have 2 providers <-
18:06.52leifmadsenmy work here is done :)
18:07.25dshap[TK]D-Fender: good point, i should try to see if the problem is mine or my provider's
18:07.49DarthPointer<carrar> <---- Radioactive
18:09.02ricko73leifmadsen: you don't expect a check in the mail for this work do you?
18:09.10leifmadsenricko73: invoice has already been sent
18:09.22leifmadsenricko73: s/check/cheque/
18:09.28ricko73bah
18:09.32ricko73beer/bier
18:09.36leifmadsentotally
18:09.40leifmadsens/soccer/football/
18:09.48carrarI may be
18:09.56carrarwe'll see, should be fun
18:10.03ricko73hockey/hockey,eh/
18:12.21dshap[TK]D-Fender: each of your suspicions are correct.  i just tried switching providers (all i changed was the Dial() command) and then i got it to work
18:12.30dshap[TK]D-Fender: further, it only worked on my landline!
18:14.05dshap[TK]D-Fender: to be honest, i don't care if i get all of this working correctly.  all i want to do is to be able to SendDTMF() for an outgoing call so i'm gonna try my .call file with flowroute now and see if it works
18:19.21*** join/#asterisk tamiel (n=tamiel@ip-120.net-81-220-18.versailles.rev.numericable.fr)
18:20.49dshapstill can't get this to work...if i can send the DTMF tones through my * box and interact with an IVR as i described earlier, does that necessarily mean SendDTMF() will work the same way?
18:21.08dshapi tried putting in a second argument for 500 ms between tones
18:21.09dshapstill nothin
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18:23.39*** join/#asterisk lanning (n=lanning@nat/yahoo/x-b930cba2686976ea)
18:26.21dshapbut i'm skeptical of flowroute
18:26.42dshapwhen i call my flowroute DID, i can't even hear the Playback()s from my * box
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18:31.15dshap[TK]D-Fender: i think i'm really just not familiar enough with Asterisk/VoIP yet to troubleshoot this on my own.  what would be your next test if you were trying to get this working?
18:32.13carrardshap, did you read leif's book?
18:32.18carrarcover to cover
18:32.43[TK]D-Fenderdshap: You've never shown us anything.  Your descriptions have more holes than swiss cheese.  You complicate your tests and then make figuring out even what you're attempting exceedingly difficult.
18:32.54carrarmmm swiss cheese
18:33.08dshapcarrar: not cover to cover but i'm working on it
18:33.08*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:34.03carrarmelted swiss cheese on toast with BEEF
18:34.03carrarand onions
18:34.05DarthPointerdshap: post your configs over at http://pastebin.ca/ ; I've been lurking on your problem, but just test it out one piece at a time to figure out where it's broken
18:35.05dshap[TK]D-Fender: this is the 100%, crystal clear, no bullshit description of what i'm trying to do right now: I want a call file on my * box to place an outgoing call to my AT&T mobile voicemail backdoor number.  I want my * box to SendDTMF() my mobile phone number so that the AT&T system will transfer it to my voicemail.  Then i want my * box to play a sound and leave me a voicemail and hang up
18:35.13dshapDarthPointer: i'll get it right up
18:36.29dshaphere's my sip.conf http://www.pastebin.ca/1446576
18:38.49dshaphere's my call file and the extension that it triggers: http://www.pastebin.ca/1446578
18:41.49dshap[TK]D-Fender: i agree with what you said.  how can i expect anyone to effectively help me if i don't get them the info they need?  that's my bad and i apologize.  i really do appreciate everyone's help here
18:44.06[TK]D-Fenderdshap: Have it call your cell.  Watch the call (FFS PB IT),  Listen for the DTMF yourself
18:44.36ariel_dshap: have you created a local IVR for testing?  You can also send your inbound call to echo your touch tones.  Since your provider is using ulaw you also might want to try dtmf=inband.
18:44.58*** part/#asterisk enzo (n=enzo@extranet.source-rh.com)
18:45.06dshap[TK]D-Fender: okay i'll try this.  i thought i did before and did some research and found that the only DTMF tones you can actually hear on the channel are "inband" tones which 'im not set up for
18:45.30[TK]D-Fenderdshap: You read wrong.
18:45.43dshapso i should be able to hear the rfc2833 signaling then?
18:45.49[TK]D-Fenderdshap: JUST DO IT
18:45.55dshapdoing it now
18:46.13*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
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18:50.06dshap[TK]D-Fender: i've verified that i cannot hear the tones when i have it call my cell phone
18:50.35[TK]D-Fenderdshap: .............
18:51.00dshap[TK]D-Fender: i get the call, pick up, see "SendDTMF" show up on my CLI, and i don't hear anything
18:51.14dshapoh
18:51.15dshapPB it
18:51.17dshapdoing that now
18:52.58*** join/#asterisk nullable_type (n=nullable@hq.verbx.net)
18:52.59dshap[TK]D-Fender: here it is: http://pastebin.com/d2628035f
18:53.19dshap[TK]D-Fender: but i also didn't hear the hello-world from the Playback(), which is an issue i've yet to resolve with Flowroute
18:53.56[TK]D-Fenderdshap: cOMPLETE LACK OF AUDIO IS THE PROBLEM HERE.
18:54.19[TK]D-Fenderdshap: Nice to find this out NOW. After wasting all our time concentrating on **DTMF**
18:54.46[TK]D-Fender"Oh wait, you mean I should try to start the car before testing coolant pressure?!?
18:55.04dshap[TK]D-Fender: we ditched the voipms issue earlier though
18:55.15dshapaudio works with voipms but DTMF does not
18:55.26dshapDTMF sort of works with flowroute but audio does not
18:55.41[TK]D-Fenderdshap: you keep pulling these stupid surprises that waste our time.  I'm tired of the runaround
18:55.42dshapwhen i say sort of works, i mean it will pass the DTMF through my * box whem my * box is connecting 2 PSTN lines
18:56.17*** join/#asterisk Winkie (n=urmom@ur.fa.gs)
18:56.19dshapi said this earlier: <dshap> when i call my flowroute DID, i can't even hear the Playback()s from my * box
18:56.33dshapand the reason i said that is because i thought it was relevant
18:56.59nullable_typeHey guys, I am having problem detecting what user is entering via the WaitExten() function, it doesn't detect and timesout. I have tried different dfmtmode in sip.conf with no success. Any suggestions?
18:57.59[TK]D-Fendernullable_type: tell us what device you are testing, show us the call, and show us your configs.
18:58.23dshap[TK]D-Fender: flowroute *successfully* transferred DTMF from my cell phone, through my asterisk box, to the AT&T voicemail system
18:58.30dshap[TK]D-Fender: voipms failed to do this
18:58.45dshap[TK]D-Fender: voipms successfully plays audio using Playback()
18:58.53dshapand flowroute fails to do that
18:59.01[TK]D-Fenderdshap: Do an ANSWER before your WAIT
18:59.22dshapok
19:00.31*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
19:01.28dshapjust tested that by having * call my cell phone with flowroute, i heard nothing at all (no DTMF or audio)
19:01.57[TK]D-Fenderdshap: Did you see the dialplan executing only AFTER you actually picked up?
19:01.58dshapnow trying voipms
19:02.28dshap[TK]D-Fender: yes that is correct.  before i picked up it just said "Attempting call" on the CLI
19:02.57dshapi just tested the exact same thing with voipms and i didn't hear anything for the DTMF but i did hear the hello-world Playback()
19:04.03dshap[TK]D-Fender: i e-mailed the Voip.ms support and they e-mailed me back saying that i shouldn't actually hear the DTMF tones since they are not "inband" - i know you said this is wrong but i just wanted to let you know that they said this
19:04.45[TK]D-Fenderdshap: Does the recording work?
19:04.57Qwell[TK]D-Fender: NAT.
19:05.02Qwellpromise.
19:05.04[TK]D-Fenderdshap: And do "playback(silence/2) after your Answer
19:05.30dshap[TK]D-Fender: what do you mean the recording?  okay i'm putting playback(silence/2) in right now after Answer
19:08.04[TK]D-FenderQwell: Well we did beat this one over him about a dozen times, and the configs do look right
19:08.29Qwelldoes externip still end in .1?
19:08.42[TK]D-FenderQwell: externip=70.181.88.1
19:08.49Qwellwinks
19:09.02nullable_typeD-Fender: RE:  problems detecting digits. My Sip, Extensions(Partial), CLI log etc are here: http://pastebin.com/d253fff8e
19:09.21*** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
19:09.21Qwelldshap: tell him your IP.
19:09.40dshap[TK]D-Fender: silence didn't make a difference for either provider on the DTMF and it didn't make a difference for the flowroute audio
19:09.41dshapalso
19:09.45[TK]D-Fendernullable_type : exten => GetDigitToConfirm,n,WaitExten() ;THIS WORKED WELL WITH EVERY OTHER PROVIDER. <- not going to F-ING trust
19:09.45dshapmy externip is not my IP
19:09.52dshapit's my router's default gateway address
19:10.01dshapthat's the only way i could get my outgoing calls to work
19:10.02[TK]D-Fenderdshap: TWIT!
19:10.07defsdoorheh ?
19:10.10[TK]D-Fenderdshap: has to be its F-ING ADDRESS
19:10.31[TK]D-Fenderdshap: * is not asking for your NEIGHBOUR'S address, its asking for YOURS
19:10.34defsdoorso your other end is talking to your default gateway - not an machine you own ?
19:10.35dshapif i set externip=MyIP (which is DIFFERENT from my router's default gateway), then outgoing calls do not go through
19:10.45dshapthat's when i get the circuits busy thing
19:10.57Qwelldshap: we went over this last night...
19:11.05[TK]D-Fenderdshap: your router's gateway is where IT sends traffic to, but that is not ITS IP address
19:11.17defsdoordshap: does your router have an internet routable IP address ?
19:11.25Qwelldefsdoor: it does.
19:11.28dshapyes
19:11.31defsdooruse that
19:11.38defsdoornothing else will work
19:11.40[TK]D-Fendernullable_type: -- Executing [Confirm@tpc:4] WaitExten("SIP/mytel-b6409920", "") in new stack <-- doesn't match
19:11.45dshapim telling u if i set it to that i'll get the circuits busy error
19:11.50[TK]D-Fendernullable_type: Now don't waste my time showing me partial crap
19:11.55defsdoorit's like sending a letter with the post office for the reply to address
19:12.08[TK]D-Fenderdefsdoor: Perfect description
19:12.10dshapdefsdoor: i understand the concept of my router's WAN IP and Default Gateway
19:12.26dshapdefsdoor: what i don't understand is why Asterisk only can place outgoing calls when i use the gateway number
19:12.30[TK]D-Fenderdshap: yo need to put your router's WAN IP as "externip"
19:12.41dshapdefsdoor: i changed it to the gateway address in an act of absolute desperation when nothing would work, and then it worked
19:12.42defsdoordshap: so - no matter what you might think could work or should work - the only thing that /will/ work is the address of your external facing IP address
19:12.43ricko73defsdoor: I had a neighbor send a postcard from Europe to "the corner of Gass Lake and Clover Rd" with the city and it got there
19:13.15defsdoordshap: and then configure your router to forward rtp to your internal address - no masqing
19:13.31dshap~rtp
19:13.32infobotrumour has it, rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
19:14.01dshapdefsdoor: if i set a DMZ on my internal address, would that effectively do what you just said?
19:14.09defsdoorwhat ?
19:14.17Qwelldshap: Don't do that.
19:14.23nullable_typeD-Fender >> You sound like a fucking retarded faking here you want to help, but all you want to do is bullshit and insult others
19:14.28[TK]D-FenderQwell: It should at least work...
19:14.47dshapdefsdoor: sorry forget that, i'm just trying to figure out how i should configure my router to forward rtp to the internal address
19:14.53[TK]D-Fendernullable_type: You show me 2 things that don't match and only little bits.  Do you really want help?
19:15.06nullable_typeD-Fender >> Now go fuck yourself, stop pretending like you want to help people
19:15.27defsdoorooo nice
19:15.36dshap[TK]D-Fender & Qwell: you're telling me that you think my DTMF issue is due to my externip setting even though i can make outgoing calls and receive incoming calls with my current setting?
19:15.43dshap& defsdoor*
19:15.44Qwelldshap: Yes.
19:15.53*** part/#asterisk nullable_type (n=nullable@hq.verbx.net)
19:16.03ricko73bye bye
19:16.08dshapQwell: okay understood.  i am now changing externip=my actual IP from whatismyip.com
19:16.08[TK]D-Fenderdshap: get it right or things will half-work at best, which BTW is what you're seeing here already.
19:16.20dshap[TK]D-fender: true
19:16.27Qwellwhat he said.
19:17.27[TK]D-FenderNobody ever listens...
19:17.41dshapokay back to square 1 then
19:17.43dshap<PROTECTED>
19:17.43dshap[Jun  3 12:17:34] NOTICE[17570]: chan_sip.c:2953 auto_congest: Auto-congesting SIP/flowroute-0a44d180
19:17.43dshap[Jun  3 12:17:34] NOTICE[17663]: pbx_spool.c:355 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
19:17.51*** join/#asterisk InHisName (n=InHisNam@68.80.23.194)
19:17.52[TK]D-FenderQwell: Thanks for the added insight on the IP itself... did look kinda legit, and .1 can be legal...
19:17.53dshapi'm assuming u want the SIP debug on the call attempt?
19:18.06[TK]D-Fenderdshap: of course
19:18.15defsdoordshap: what is your * box's default gateway ?
19:18.34defsdoordshap: it /is/ the internal address of your router isn't it ?
19:18.48dshapyes i believe that is correct
19:18.54defsdoorcheck it
19:19.02dshapim not sure how to do it in linux
19:19.09[TK]D-Fenderdefsdoor: that is fine... otherwise the packets wouldn't be getting out...
19:19.14defsdoornetstat -r
19:19.19[TK]D-Fenderdshap: "rounte -n"
19:19.23[TK]D-Fenderroute*
19:19.30defsdoor[TK]D-Fender: depends - he might have some RIP nonsense and other route out
19:19.40InHisNameI have phrase: exten =>s,1,NoOp(Exten=${EXTEN:2})   and it dials 3rd digit to end of string
19:19.56*** join/#asterisk nullable_type (n=nullable@hq.verbx.net)
19:19.58InHisNameI would like to dial 3rd digit thru 12th digit
19:20.01dshap[root@localhost asterisk]# route -n
19:20.01dshapKernel IP routing table
19:20.01dshapDestination     Gateway         Genmask         Flags Metric Ref    Use Iface
19:20.01dshap192.168.2.0     0.0.0.0         255.255.255.0   U     0      0        0 eth0
19:20.01dshap169.254.0.0     0.0.0.0         255.255.0.0     U     0      0        0 eth0
19:20.02dshap0.0.0.0         192.168.2.1     0.0.0.0         UG    0      0        0 eth0
19:20.03[TK]D-FenderInHisName: that doesn't dial anything
19:20.04dshap[root@localhost asterisk]# netstat -r
19:20.06dshapKernel IP routing table
19:20.08dshapDestination     Gateway         Genmask         Flags   MSS Window  irtt Iface
19:20.10dshap192.168.2.0     *               255.255.255.0   U         0 0          0 eth0
19:20.12dshap169.254.0.0     *               255.255.0.0     U         0 0          0 eth0
19:20.14dshapdefault         192.168.2.1     0.0.0.0         UG        0 0          0 eth0
19:20.17[TK]D-Fenderdshap: dshap STOp SPAMMING
19:20.21dshapsorry
19:20.24dshapi'll PB that
19:20.25ricko73~pastbin
19:20.28[TK]D-FenderInHisName: [15:20]<dshap>0.0.0.0 192.168.2.1 0.0.0.0 UG 0 0  0 eth0 <--- your gateway
19:20.29ricko73~pastebin
19:20.30infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:20.32defsdoortoo late now
19:20.32[TK]D-Fender(G)
19:20.38dshapk
19:20.43dshapright
19:20.43kaldemarInHisName: ${EXTEN:2:10}
19:20.45dshapthat's what i thought
19:20.45[TK]D-Fenderdshap: : [15:20]<dshap>0.0.0.0 192.168.2.1 0.0.0.0 UG 0 0  0 eth0 <--- your gateway
19:20.48dshap192.168.2.1
19:20.50*** part/#asterisk nullable_type (n=nullable@hq.verbx.net)
19:21.18InHisNamethanks kaldemar, that should work. I'll try now.
19:21.47defsdoordshap: just out of interest what is the router ?  I've have some trouble with some that think they know how to manage SIP and come with a preset ruleset for it (that doesnt work)
19:22.26Qwelldefsdoor: You just described *every* "sip aware" router.
19:22.32defsdoor:)
19:22.46dshapdefsdoor: Belkin F5D8230-4 v2
19:22.49dshapit's a POS
19:23.16dshap[TK]D-Fender: pastebin of the outgoing call attempt with my externip reset to my actual IP address: http://pastebin.com/m71cc818c
19:23.37*** join/#asterisk hepta (i=cso@78.156.12.251)
19:24.34InHisNameI thought this was parse string of 12 or more digits:   91XXXXXXXXXX.   with . meaning "or more"    Correct me if wrong.....
19:25.12kaldemarInHisName: . means one or more characters
19:25.22*** join/#asterisk |Cybex| (n=John@80.100.126.176)
19:25.30[TK]D-FenderInHisName: No, that is 12 digits + 1 or more CHARACters (13 MIN total)
19:25.40InHisNamekaldemar:  so I typed 13 or more dig by accident ?
19:25.48heptai have two video phones that i can communicate with directly.  i register them on asterisk and let one call the other using Dial.  asterisk only does audio.  show codecs lists video codecs that i support.  i have enablevideo=true in sip.conf and wonder how i make asterisk bridge my video calls.
19:26.34[TK]D-Fenderhepta: Pick that actual video codecs as well
19:27.00heptasorry, i didnt understand
19:27.46heptai have all the 26x video codecs in the devices
19:28.11[TK]D-Fenderhepta: in ASTERISK
19:28.27heptawhat about it
19:28.44*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
19:28.57[TK]D-Fenderhepta: specify your VIDEO CODECS in your PEER entries in ASTERISK or it will only offer AUDIO]\
19:29.11heptaoh, as allow=?
19:29.14[TK]D-FenderYES
19:29.27heptathanks.  ill try that
19:29.28defsdoorallow=g729,gsm,ulaw,alaw,h263
19:29.43defsdoorthat's what I have for my n8[01]0
19:29.45heptaso default allow is all the audio codecs of *?
19:32.09[TK]D-Fenderhepta: There is no such thing as "default"
19:32.19*** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com)
19:32.39heptawhat if i have al
19:32.48heptaallow=all
19:33.16*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:33.36*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
19:34.13dshapanyone pick up any clues from that SIP DEBUG by any chance?
19:34.33*** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com)
19:34.36eppigyhello
19:34.38eppigyi am dave
19:36.39defsdoor> Reliably Transmitting (no NAT) to 70.167.153.130:5060  is that correct ?
19:37.52[TK]D-Fendereppigy: We could see that :)
19:39.19dshapdefsdoor: could be wrong, but i think that's because i have nat=no under my [flowroute] in sip.conf.  that is flowroute's IP.  i read the NAT guide that the infobot has and it says that if you are connecting to someone who's not behind a NAT (i.e. my provider), then you put nat=no there
19:39.39dshap~nat
19:39.40infobotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
19:39.40dshapoh
19:39.42dshap~sipnat
19:39.43infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
19:40.50defsdooryou got nat=yes in general though ?
19:40.54dshapyes
19:41.15dshapdshap> here's my sip.conf http://www.pastebin.ca/1446576
19:41.31infernixQwell: tried chan_mobile with android v1.5 yet?
19:42.31*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
19:42.44dshap[TK]D-Fender: anything else you'd like me to try?
19:44.20dshapi thought it was promising before when i had my other externip setting.  i could do just about anything.  even transmit DTMF through my provider.  only thing that wouldn't work is SendDTMF and i'm just having a hard time understanding why that's related to my network setup
19:47.35jayteeTRABAJO!
19:49.26*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
19:50.18dshapim going to go grab something to eat but im gonna stay in the channel so i can receive any messages that anyone might want to send when i am gone
19:50.20dshapi'll be back later
19:50.24dshapthanks again for the help so far
19:51.10defsdoor# SIP/2.0 407 Proxy Authentication Required
19:56.37defsdoordoes he need to register ?
19:58.03[TK]D-Fenderdefsdoor: No.
19:58.08[TK]D-Fenderdefsdoor: Just basi auth
19:58.16defsdoorit doesnt seem to be doing it
19:58.39[TK]D-Fenderdefsdoor: the "retransmit" bits indicate something else is wrong...
19:59.28defsdooroh I see why
19:59.35*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
19:59.49*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)
20:02.51defsdoorare the call ids ok ? different addresses on the xmit and rcv
20:03.58*** join/#asterisk aksyn (n=aksyn@gw.na.nu)
20:09.08*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
20:24.00*** join/#asterisk dexthageek (n=mike@66.134.255.227)
20:24.07dexthageekgood afternoon
20:24.41dexthageekI am running into a problem with Asterisk 1.6 crashing when more then one person enters a meetme conference
20:25.12dexthageek1st person enters and waits with MOH
20:25.24dexthageek2nd person logs in and asterisk crashes
20:25.44*** join/#asterisk acxty (n=acxty@201.220.136.117)
20:27.34dexthageekanyone around?
20:28.25*** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
20:28.45*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
20:28.48Katty:>
20:28.51Katty:>>>
20:29.26acxtyHi guys, I am testing the call files right now. I want to make a call to a cellphone this is what I have http://dpaste.com/51159/
20:29.34acxtyit is returning some errors
20:30.03dexthageekit is quiet in here, I have been waiting for a response myself
20:30.26[TK]D-Fenderacxty: Channel: SIP/504111111 <--- nothing to call.  guess you don't have a [504111111] in sip.conf
20:31.55[TK]D-Fendercheckout time, BBIAB
20:32.16eppigyhello Katty
20:32.19eppigy:>
20:32.30dexthageekI am running into a problem with Asterisk 1.6 crashing when more then one person enters a meetme conference
20:33.55*** part/#asterisk aksyn (n=aksyn@gw.na.nu)
20:34.18jayteeKatty !!!!! *hugs*
20:34.50juanIMPwhats the cli's output dexthageek ? have you check full messages or debug?
20:35.06acxty[TK]D-Fender, I correct that part but now I am getting this http://dpaste.com/51161/
20:35.15*** join/#asterisk bbryant1 (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
20:35.17KattyHAI DAVE! :>
20:35.20Kattyhugs eppigy
20:35.26Kattyhugs on jaytee
20:35.31dexthageekyup nothing coming up in full or messages
20:35.36dexthageekast just core dumps
20:35.53jayteeacxty, [TK]D-Fender has left the building
20:36.12jayteedave, where've ya been man? on vacation?
20:36.24acxtyjaytee, may you help me with the call file?
20:37.24jayteeacxty, nope. not my bailiwick, cup of tea, strongsuit, etc.
20:37.28juanIMPwhile the second guy enters to the conference .... it just crashes?
20:38.51dexthageekyup, safe_asterisk kicks in and restarts
20:39.20jayteetiming source is? hardware, dahdi_dummy?
20:39.39dexthageekdahdi_dummy
20:40.06jayteewhich version of 1.6.x? and what version kernel?
20:40.32juanIMPdmesg shows something wrong while dahdi_dummy is on ?
20:41.35beekacxty: Specify which context you want that to drop into.
20:44.03dexthageek1.6.0.5
20:44.11dexthageekwe are running on EC2
20:45.20dexthageekunfortunately we cannot upgrade to the latest release
20:47.00leifmadsendexthageek: not much you're going to be able to figure out without a backtrace and an issue report
20:47.19leifmadsensee doc/backtrace.txt in your asterisk source
20:47.44jayteeEC2? Amazon's cloud?
20:48.46*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
20:48.56*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
20:51.20dexthageekjaytee: yes
20:51.59jaytee"For the love of God, Montressor!!!"
20:53.12jayteedexthageek, is this a new install and a test platform? or have you had this working already with a previous version of * running on EC2?
20:54.11dexthageekthis is our first install of Asterisk 1.6
20:54.35dexthageekwe have 6 ec2 instances running ast 1.4
20:55.06jayteedexthageek, running zaptel with ztdummy or dahdi with dahdi_dummy?
20:55.18dshapalright new potential plan: what if i set my externip back to the gateway address so i can make outgoing calls again and then troubleshoot the flowroute audio issue?  i already confirmed that flowroute can pass DTMF in & out of its system through my * box so maybe if i get the audio to work on flowroute then SendDTMF() might work as well.  thoughts?
20:57.38dshapthe fact that Playback() doesn't actually send audio over flowroute for my * box but it works fine with my other provider, voipms
21:01.06*** join/#asterisk telecos (n=sergio@210.167.219.87.dynamic.jazztel.es)
21:03.43*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:04.46dshap[TK]D-Fender: new potential plan: what if i set my externip back to the gateway address so i can make outgoing calls again and then troubleshoot the flowroute audio issue?  i already confirmed that flowroute can pass DTMF in & out of its system through my * box so maybe if i get the audio to work on flowroute then SendDTMF() might work as well.  thoughts?
21:05.16*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
21:06.07*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
21:06.07*** mode/#asterisk [+o leifmadsen] by ChanServ
21:06.16[TK]D-Fenderdshap: Don't screw with your networking, and do it like you've been told.  Next IMAGEBIN.CA your route config pages so we can see you've done things right.  Next PB your configs & new call attemps.  Pick on provider and beat them to deatch till they work before even TALKING about the other
21:06.49dshapcoming right up
21:07.24eppigyKatty: whats for din din
21:07.28eppigy8[]
21:07.50eppigyjaytee: my shell server got rebooted
21:07.58eppigyit takes me time to reconnect to every network
21:07.59eppigy:[
21:09.13*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
21:12.04*** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com)
21:12.50thomashello
21:12.57dshap[TK]D-Fender: here is what you requested
21:13.01dshapRouter config: http://imagebin.ca/view/4Krlzg2.html
21:13.02dshapsip.conf: http://pastebin.com/m84f6426
21:13.02dshapCall attempt (.call file) SIP DEBUG: http://pastebin.com/m528bcfb2
21:13.10*** part/#asterisk lanning (n=lanning@nat/yahoo/x-b930cba2686976ea)
21:13.11dshapin retrospect i probably should have made that router config private content but whatever
21:13.32thomasi have a little problem
21:14.15thomashave in my dialplan:
21:14.15thomasexten => _fromAmtX.,1,Dial(SIP/toAnlage${EXTEN:7}@berofix)
21:14.15thomasexten => _fromA[n]lageX.,1,Dial(SIP/toAmt${EXTEN:10}@berofix)
21:17.30thomasWhen i calling my number and i have a source-number then ringing my phone (all ok): http://paste.keks.be/500/txt
21:17.50thomaswhen i calling my number and i havent a source-number then:
21:17.50thomas[Jun  3 23:16:40] NOTICE[2630]: chan_sip.c:13879 handle_request_invite: Call from '' to extension 'fromAmt66668269' rejected because extension not found.
21:18.08thomasany ideas to my problem?
21:18.20thomasmy dialplan:
21:18.20thomasexten => _fromAmtX.,1,Dial(SIP/toAnlage${EXTEN:7}@berofix)
21:18.21thomasexten => _fromA[n]lageX.,1,Dial(SIP/toAmt${EXTEN:10}@berofix)
21:19.07dshap[TK]D-Fender: this also: http://imagebin.ca/view/dJB5nN.html
21:20.45[TK]D-Fenderdshap: .....
21:20.53[TK]D-Fenderdshap: WTF is RTP?!
21:20.53dshapwhat else do you need?
21:21.02thomas[TK]D-Fender: hello. excuse me, but, can you help me with my problem?
21:21.10dshap~rtp
21:21.11infobot[rtp] The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets.
21:21.22dshapokay so i need to forward another port is what you're saying
21:21.24[TK]D-Fenderdshap: WHERE the &#^$ is it?
21:21.34[TK]D-Fenderdshap: You didn't forward the ports for RTP <_
21:21.35dshapi dunno
21:21.41[TK]D-Fender.........
21:21.41eppigyoh boy
21:21.48dshapokay im sorry RTP is a new concept that i just learned today
21:21.48[TK]D-Fenderreaches for his ClueBat (tm)
21:21.49dshaphah
21:21.54dshapwhat ports?
21:22.01[TK]D-Fender~sipnat
21:22.02infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:22.03[TK]D-Fender^^^^^^^^^^^^
21:22.38dshaphttp://www.voip-info.org/wiki/view/RTP+Ports
21:22.44dshapRTP: UDP ports 16384-32767
21:22.46dshapi should probably forward those
21:23.03[TK]D-Fenderdshap: I've linked you the guide about a dozen times.  READ THAT ONE
21:23.09*** join/#asterisk joako (n=joako@opensuse/member/joak0)
21:24.24dshapi've read that guide
21:24.27dshapi think i'm #3
21:24.28Kattyeppigy: idunno what's for dindin :<
21:24.37Kattyeppigy: that requires work. i'm all worked out
21:24.49dshapthe guide links to a tip that says if i don't add "qualify=yes" then I  won't be able to receive SIP calls, which is obviously wrong
21:24.59[TK]D-Fenderdshap: 1st f-ing link
21:25.26dshapshit i think i overread the RTP thing on that first link
21:25.28dshapughh
21:25.55thomascan anyone help me ?
21:26.26jayteeeppigy, I missed ya man! you've been gone for days seems like
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21:27.08eppigy:D
21:27.14eppigyyes it has been a minute
21:27.25eppigyKatty: well I will make din din then
21:27.40eppigyplots a course to Boston Market
21:27.44Kattyeppigy: k
21:28.58dshap[TK]D-Fender: for some reason my router needs to restart to apply port forward changes
21:28.59dshapbrb
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21:29.14jayteeI liked it better when it was Boston Chicken and it wasn't national. they only had like 3 places in the Boston area. food was better then.
21:30.16jayteeanyway, it's quittin time. bbiab
21:30.48AlmightyOatmeal[Jun  3 16:29:48] NOTICE[57548]: chan_sip.c:14736 handle_request_invite: Call from '<phone number>' to extension '<phone number>' rejected because extension not found. <-- uh oh?
21:31.36AlmightyOatmealwhen i registered my asterisk box on my sip provider it told me to specify an extension like 100, which i did which should ring to mmy phone..
21:31.47AlmightyOatmeali'm not really sure what i didn't do right :(
21:32.04jayteeever wonder what would happen if you dialed a 4 digit extension that you knew didn't exist in your dialplan or wasn't reachable from the context of the device you were calling from? :-)
21:32.39AlmightyOatmealwas that geared toward me?
21:33.02jayteeyes, MrOmnipotentHighFiberCereal
21:33.45AlmightyOatmealjaytee: thats an incoming call from a landline though.. i thought it would bring up a menu prompt or ring directly to extension 100 :(
21:34.06Kattyyawns sleepily
21:34.17Kattyfalls asleep somewhere, randomly
21:34.30jayteehmm, well I'm on my way out or I'd look at your dialplan. maybe later if you're still stuck and no else has managed to help you sort it out.
21:34.43AlmightyOatmeali guess i need to work on my dialplans :)
21:34.47AlmightyOatmealjaytee: i work third shift so i'll be on definetly when i get to work :)
21:35.19*** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net)
21:35.31dshap[TK]D-Fender: now I have the entire RTP port range forwarded to my * box
21:35.47jayteeand excuse the offhand responses, my buttocks doth contain great wisdom that verily I oft find impossible to contain! :-)
21:35.50dshapdid not fix my problem
21:36.16jayteein simple english, I'm a wiseass!
21:36.28jayteeback later all
21:36.30jayteepeace
21:37.00AlmightyOatmeallol
21:38.11thomasIf i calling with my phone and send my phonenumber (sourcenumber), then I have no problem, phone is ringing. If I call without send my phonenumber (source) then i have a problem:
21:38.25thomasNOTICE[2630]: chan_sip.c:13879 handle_request_invite: Call from '' to extension 'fromAmt66668269' rejected because extension not found.
21:38.28thomasany ideas?
21:39.39*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
21:39.41acxtybeek, I want to make a outgoing call, specifically a cellphone
21:40.51lanceythomas: you sure you have such extension called 'fromAmt66668269' o_O?
21:41.46acxty[TK]D-Fender, I fix what you told me earlier. I am getting this error know http://dpaste.com/51161/
21:41.53thomaslancey: when i have set a sourcenumber. http://paste.keks.be/501/txt
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21:42.56lanceythis seems to be calling different extension?
21:43.15[TK]D-Fenderacxty: and I don't see SIP debug in there, and no configs
21:43.39[TK]D-Fenderdshap: "type=peer", remove the "insecure" line.
21:43.55acxty[TK]D-Fender, only sip.conf or something else?
21:44.08[TK]D-Fenderacxty: sip.conf
21:44.16afinkguys I just moved an asterisk box and I get this when trying to start zaptel:  http://pastebin.com/m3b672fda
21:44.27acxtyok
21:44.40AlmightyOatmealwhen i recieve an incoming call it tries to connect to an extension umber thats the same as the number for the box.. is this normal or do i need to create a dialplan for it or something?
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21:45.21[TK]D-FenderAlmightyOatmeal: "the number for the box"?  Huh?
21:45.51AlmightyOatmeal[TK]D-Fender: [Jun  3 16:43:43] NOTICE[57548]: chan_sip.c:14736 handle_request_invite: Call from '6088074038' to extension '6088074038' rejected because extension not found.
21:46.05acxty[TK]D-Fender, here it is http://dpaste.com/51187/
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21:46.42dshapthat was weird - my router just dropped wifi
21:46.45dshapim back now though
21:46.47dshap*sigh*
21:47.38AlmightyOatmeal[TK]D-Fender: what did i screw up? heh :(
21:48.21[TK]D-FenderAlmightyOatmeal: Its looking for the exten you see there in a certain context SIP DEBUG would reveal.... and you don't have a match
21:48.55[TK]D-Fenderdshap: JUST fyi, YOUR ROUTER, BEING THE pos THAT IT IS, COULD BE THE ENTIRE CULPRIT HERE.
21:49.26AlmightyOatmeal[TK]D-Fender: i guessed that, but i didn't realize i had to have an extension that matched my box's phone number.. to me that doesn't look normal
21:49.27dshap[TK]D-Fender: later im going to connect my server directly to my modem and see if i can send the DTMF
21:50.13acxty[TK]D-Fender, do you found something wrong on the sip.conf file?
21:51.11[TK]D-FenderAlmightyOatmeal: when the call comes in, it would be natural that it would be a call TO the number you are being provided.  Who it is FROM is the CALLERID
21:51.50[TK]D-Fenderacxty: Your call file is not calling a target NUMBER there
21:52.30AlmightyOatmeal[TK]D-Fender: so i need to create an extension with the phone number? hum ok
21:53.04acxty[TK]D-Fender, I have Extension: on it
21:53.10[TK]D-FenderactShow me
21:53.21acxtydo I need to add that extension on the sip.conf?
21:53.42[TK]D-FenderactNO.  the CHANNEL does not tell your PROVIDER what # to call.  Or is that some fixed enpoint like a SIP phone?
21:54.13acxty[TK]D-Fender, no, pure asterisk
21:54.22[TK]D-Fenderacxty: Who are you calling?
21:54.36acxtyI am trying to make a call from asterisk to a cellphone
21:55.00[TK]D-Fenderacxty: What cellphone?
21:55.09[TK]D-Fenderacxty: I don't see another cell # in there
21:55.09acxtymy
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21:56.12AlmightyOatmeali can't set an extension as long as my phone number ugh.. i dont get it
21:56.19acxty[TK]D-Fender, I thoug that using Extension: will do the call, Am I wrong?
21:57.21*** join/#asterisk scooby2 (n=scooby2@pdpc/supporter/active/scooby2)
21:58.17[TK]D-Fenderacxty: Yes, you are.  you nned your Channel: line to look EXACLY like you would if you used in in a DIAL() command
21:58.33[TK]D-Fenderit*
21:58.45[TK]D-FenderAlmightyOatmeal: Pardon?  Why the hell not?
21:58.49acxty[TK]D-Fender, SIP/50411111/99999999
21:59.16[TK]D-Fenderacxty: Yes, that looks more normal
21:59.23AlmightyOatmeal[TK]D-Fender: i dont get why i dont get the phone directory upon calling the number, i dont get why the error happens and i dont know enough to fix it yet
21:59.30AlmightyOatmealits a little frustrating for me until i learn more
22:00.32scooby2Is there a way to log the number someone called in the CDR?
22:00.48scooby2I strangely remember this working in 1.2 but I just noticed in 1.4 it is not.
22:00.52dshap[TK]D-Fender: the RTP port forwarding made SendDTMF() work with my voip.ms provider!!!!
22:00.58acxty[TK]D-Fender, if I change that, how does the Extension: will go?
22:04.00acxty[TK]D-Fender, I set extension 0, it is working know thanks ;)
22:04.57AlmightyOatmealso i can't make a call either:
22:05.02AlmightyOatmeal[Jun  3 17:04:39] NOTICE[57548]: chan_sip.c:14736 handle_request_invite: Call from 'jamie' to extension '6083994252' rejected because extension not found.
22:05.06AlmightyOatmeali just don't understand
22:05.17scooby2I see the problem. For some reason src and dst in cdr are being input into mysql as both being the src #
22:06.09dshap[TK]D-Fender: I'll deal with flowroute later.  for now I'm going to continue development on my project with voip.ms.  thank you very much for helping me resolve the problem.  i will probably be back in this channel another time for other issues that i come across as I continue development
22:06.26dshap[TK]D-Fender: although the externip issue remains a mystery
22:06.28[TK]D-Fenderdshap: I was afraid of that ;)
22:06.29lanceyAlmightyOatmeal: in your sip.conf/iax.conf, wherever you defined 'jamie' you have also specified a context
22:06.34dshaphaha
22:06.38lanceyyou don't have a '60blabla' extension in it.
22:06.54lanceyor anything else that would match this (.e.g. _X., or 6X. etc)
22:07.39AlmightyOatmealuh, i should
22:07.42AlmightyOatmeal2 sec
22:08.09AlmightyOatmeallancey: if you don't mind, http://pastebin.ca/1446829 are the extensions i have
22:08.40lanceyand what's the context of 'jamie' ?
22:08.47[TK]D-FenderAlmightyOatmeal: Now aside from all the other stuff I'd blast you for... what in there should macth 6083994252?
22:08.57lanceywell, nothing of these matches the number you dialed - 6xxxx
22:09.14lanceyyou also have a missing bracket (>) on the last line
22:09.22AlmightyOatmeallancey: the context to jamie is my default extensions that i just pasted
22:09.25[TK]D-Fenderlancey: and the context you referred to... well.. I don't even want to think where you came up with that idea :)
22:09.37AlmightyOatmeal[TK]D-Fender: i'm going directly from what BroadVoice said :(
22:09.40lancey[TK]D-Fender?
22:10.19[TK]D-FenderAlmightyOatmeal: Forget what they say, and look at what they DO.
22:10.27lanceyscratches his head
22:10.35lancey[TK]D-Fender would you mind being more clear about that, please?
22:10.42*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
22:10.44AlmightyOatmealso i should change _1NxxNxxxxxx to my phone number?
22:11.00[TK]D-FenderAlmightyOatmeal: They are targeting a TEN DIGIT number, that starts with a "6".  First I have NO reason to trust the call is even looking at the CONTEXT those extens are in.
22:11.11lanceyAlmightyOatmeal: generally, you should have at least TWO contexts
22:11.32[TK]D-FenderAlmightyOatmeal: You are showing us an INBOUND call attempt from BV, correct?
22:11.52AlmightyOatmeal[TK]D-Fender: that last one was outbound, i pasted inbound earlier
22:11.58lancey[TK]D-Fender: lol. did he call his provider 'jamie'
22:11.59lancey:)
22:12.20[TK]D-Fenderlancey: No, thats just the CALLERID of the caller
22:13.25[TK]D-FenderAlmightyOatmeal: try some new PB's
22:13.34AlmightyOatmealPB's?
22:13.37[TK]D-Fender~pb
22:13.38infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
22:13.49AlmightyOatmealof?
22:14.25[TK]D-FenderAlmightyOatmeal: your dialplan including th HEADER for your context.  For the incoming call attempt with SIP DEBUG ENABLED.  Then  once we fix that we can look at your OUTBOUND separately
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22:18.41AlmightyOatmeal[TK]D-Fender: http://pastebin.ca/1446844  <-- that?
22:20.21[TK]D-FenderAlmightyOatmeal: go to * CLI and ENABLE SIP DEBUG "sip set debug on" and try again.
22:25.28AlmightyOatmeal[TK]D-Fender: http://pastebin.ca/1446853  <-- when i dial in
22:26.21[TK]D-FenderAlmightyOatmeal: TOO LATE.  You show it only AFTER the failure
22:26.31AlmightyOatmealuh
22:26.34[TK]D-FenderAlmightyOatmeal: ENTIRE damn call please can configs along-with
22:26.49AlmightyOatmeal..
22:27.05[TK]D-Fenders/can/and/
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22:30.39AlmightyOatmeal[TK]D-Fender: http://pastebin.com/f14e254df and configs: http://pastebin.ca/1446844
22:30.54AlmightyOatmealunless you want the whole config file
22:33.15[TK]D-FenderAlmightyOatmeal: Line #176 - Contact: <sip:100@192.168.1.50> <-- your server is not configured correctly to work behind NAT.
22:33.17[TK]D-Fender~sipnat
22:33.17infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
22:33.20VaGoNeTaS[TK]D-Fender do you know, if there is possible to make an 'Virtual Fax' by receiving the faxes straight to an email address instead of a piece of paper?
22:34.12[TK]D-FenderAlmightyOatmeal: ^^^^^ follow the guide
22:34.13VaGoNeTaSin our office we dont have analogue lines anymore, only VoIP
22:34.13AlmightyOatmeal[TK]D-Fender: ok
22:34.16VaGoNeTaSi heard something about HylaFax, , is that possible? or i still need an ATA modem or something?
22:34.57pfnI forget, what's the process for exposing a blocked callerid?  forward it to an 800 number?
22:35.10[TK]D-FenderAlmightyOatmeal: Next : line 355 - Looking for 6088074038 in default (domain 192.168.1.50) <-- its looking in [default] for that EXTEN.  We know there is no match and we have no idea what you want to DO with the call when it comes in anyway.
22:35.42[TK]D-FenderAlmightyOatmeal: And you should not have your inbound call land on the same context that has OUTBOUND extens, etc.  This is a security risk.
22:36.02[TK]D-FenderAlmightyOatmeal: that is the point of contexts, to separate things.
22:39.05AlmightyOatmealso noted
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22:40.23AlmightyOatmeal[TK]D-Fender: then i need to learn more about extensions for handling inbound calls
22:40.30AlmightyOatmealand same for outbound heh
22:40.36pfnugh, why does teliax charge for inbound calls, not cheap
22:42.24AlmightyOatmealwell outbound calls seem to work
22:42.34AlmightyOatmeali'll read more on extensions tonight
22:42.54AlmightyOatmealthanks for the guidance [TK]D-Fender
22:43.46[TK]D-FenderAlmightyOatmeal: Dialplan = 95% of *
22:44.10*** part/#asterisk juanIMP (n=Juancho@200.71.41.254)
22:44.54jayteeanyone ever tried using PC-6400 800mhz DDR2 dimms on a mobo that will only accept up to PC-5400 667mhz DDR2?
22:47.10AlmightyOatmealjaytee: normally the faster ram would simply downclock, not normally a risk to stability
22:49.22jayteethat's what I was thinking...I've run 400mhz ram on a mobo that would only clock at 333mhz with no problems. came home and this beast was locked and at the reboot got the ugly 3 beep code saying base 64K was shot. Pulled the DIMM in channel 0 slot 0 and moved the one from channel 1 dimm 0 and it booted ok but now it's only got 1GB
22:50.03scooby2[TK]D-Fender: Is there any config file that could be changed to make the cdr src and dst be the same? using cdr_addon_mysql.
22:50.07jayteeI've got a pair of 2GB 800mhz DDR2 DIMMS I've never used as I was planning on building yet another damn system but hadn't gotten another mobo yet
22:50.26jayteethink I'll give it a shot. hopefully be back in just a few :-)
23:05.27KavanSanyone have a suggestion for a good voip client for mac?
23:05.41pfnso, if teliax charges for inbound minutes does it mean they should be required to expose ani/clid?
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23:22.01seb-[TK]D-Fender: sorry my Ekiga has been so cranky....got any time @home now?
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23:52.14Typhoeus_Hello. Would anyone be able to help me with a configuration issue between an Asterisk server I am setting up and a Cisco VG200? I have outgoing calls working and I think I have the dial-peer partially working but when it dials you can't here anything.
23:52.25Typhoeus_Sorry should of used a ? :)
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23:52.55johnakabeanhey everyone; i'm having a problem with asterisk. using freepbx 2.5.1 and asterisk.1.4.22, I type amportal start
23:52.59johnakabeanasterisk starts FINE
23:53.03johnakabeanand runs for about an hour
23:53.14johnakabeanthen I get an error "automatically restarting asterisk" in console
23:53.40Typhoeus_did you look at the "full" log to see if it shows why it is restarting?
23:53.46johnakabeanone sce
23:55.04dshapHow would i initiate an outgoing call from my dialplan but have it NOT bridge the call to the incoming channel?  let's say I want to call my asterisk server, dial an extension, and then have the server place a call to someone else in the background and play an audio file without connecting me to them.  can this be done with Dial()?
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23:57.21Typhoeus_dshap: Are you trying to do like a voice broadcast to a list of numbers?
23:58.20generalhanso, i have monitor-format = MixMonitor in my queues.conf for a certain queue... the calls *are* recorded, and they do save as a single file instead of the -in and -out files, but for some reason during the mix it puts the -in portion after the -out portion, and not mixed together. anyone seen this before ?
23:58.21dshapTyphoeus_: nope, just trying to do exactly what i said - have it place a single call in the background.  it would be as if i could call into my asterisk server, press a button, and then a call file would be created and the call would be executed - but obviously that's not how i want to do it
23:58.49dshapunless that is the appropriate way to do it..

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