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| 00:26.20 | s14ck | is back (gone 01:33:39) |
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| 01:53.30 | jaytee | http://tinyurl.com/p3dwbo |
| 01:53.35 | jaytee | yum! |
| 01:53.42 | jaytee | :-) |
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| 01:55.09 | Qwell | apt-get! |
| 01:55.25 | jaytee | hehe |
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| 02:02.30 | uluatu_ | people, is this possible to devicestate not report the correct state of some channel after an attendant transfer? |
| 02:11.45 | [TK]D-Fender | jaytee: Engrish FAIL |
| 02:12.23 | jaytee | :-) |
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| 04:15.14 | jnfuller | does anyone know if there are any good forums or chat rooms for digium's free fax for asterisk> |
| 04:17.28 | jnfuller | or, failing that... if you buy a channel do you get access to any better docs/support? |
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| 04:29.51 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 04:31.02 | lanning | quick SIP question: To: format is <user>@<server>/<did> correct? |
| 04:31.45 | lanning | this is the sip packet to: field... |
| 04:32.39 | [TK]D-Fender | lanning: Not TO a user, FROM a user.. |
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| 04:42.08 | jnfuller | man do I hate peering with an acme-packet sbc |
| 04:42.54 | jnfuller | one screwup in a regex and all the calls die with 404 |
| 04:43.22 | jnfuller | but no, everything's fine on their end |
| 04:43.26 | jnfuller | *sigh* |
| 04:43.43 | jnfuller | escape + often? |
| 04:43.52 | jnfuller | sorry, I'll keep my rant to myself |
| 04:44.01 | jnfuller | just needed to vent |
| 04:44.10 | carrar | tell us how you really feel |
| 04:44.49 | [TK]D-Fender | carrar: I would, but its illegal in 17 states. |
| 04:44.50 | [TK]D-Fender | carrar : But she is SO worth it.... |
| 04:44.54 | InfoMoMo | carrar: well said :D |
| 04:45.11 | carrar | heh |
| 04:45.29 | [TK]D-Fender | carrar: And the doctors tell me feeling should return there within a week |
| 04:45.54 | carrar | Thats the good times! |
| 04:46.57 | [TK]D-Fender | WE ARE FOR GOOD TIMES! |
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| 04:50.49 | jnfuller | lanning, 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) |
| 04:51.54 | lanning | the uri directly in the sip packet... |
| 04:52.19 | jnfuller | the uri follows 3261 |
| 04:53.56 | jnfuller | http://www.ietf.org/rfc/rfc3261.txt |
| 04:54.32 | lanning | as reported in sip set debug |
| 04:54.32 | lanning | ya, just been reading up. |
| 04:54.32 | lanning | looks like the provider is handing me DIDs as separate users. |
| 04:55.50 | jnfuller | there's really no such concept as did in sip, user@host |
| 04:56.50 | florz | what's DID then, if there is no such concept in SIP?! |
| 04:57.23 | jnfuller | the did is arbitrary, it's just a text field for a user |
| 04:57.41 | jnfuller | so it's up to the ua to decide if it is valid |
| 04:58.04 | *** join/#asterisk mchou_ (n=mchou@unaffiliated/mchou) |
| 04:58.22 | florz | so you say that the DID is arbitrary and therefore is not a DID?!? |
| 04:58.33 | carrar | No, there is no spoon |
| 04:58.47 | [TK]D-Fender | hordes the cutlery drawer |
| 04:58.57 | [TK]D-Fender | MY PRECIOUS!!!!!!!!! |
| 04:58.58 | jnfuller | yes for example a provider sends foo@bar.baz |
| 04:59.11 | jnfuller | foo is just a username and could be anything |
| 04:59.27 | jnfuller | it's up to the downstream server to react |
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| 04:59.31 | carrar | Should have use mailer daemon, that guy is always emailing me |
| 04:59.38 | [TK]D-Fender | "sends"? Could someone be more vague please? |
| 05:00.17 | jnfuller | no more vague than using a PRI term like did with sip ;) |
| 05:00.34 | carrar | PRI over SIP |
| 05:00.48 | carrar | PRIoSIP |
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| 05:01.11 | k-man_ | is ther Siemens C470 ip phone any good? |
| 05:02.38 | lanning | trying to get it to not use extension s |
| 05:02.59 | florz | jnfuller: Why is "DID" a "PRI term"? Is "phone call" a "PRI term", too? |
| 05:03.33 | [TK]D-Fender | and DID is not a PRI term... |
| 05:03.48 | [TK]D-Fender | There are higher protocols at work. |
| 05:03.57 | florz | but phone call is? =:-) |
| 05:04.46 | [TK]D-Fender | lanning: If you are receiving a call targeting "s" its because you didn't tell then any other number to dial when sending you calls |
| 05:04.51 | jnfuller | all I meant was that from a sip perspective it's user@host |
| 05:04.53 | [TK]D-Fender | them* |
| 05:05.24 | florz | jnfuller: _what_ is "user@host", and what does that have to do with the definition of "DID"? |
| 05:05.52 | jnfuller | Well, what's your definition of did ? |
| 05:06.09 | florz | that was kindof the question I was asking you, I guess =:-) |
| 05:06.22 | lanning | http://pastebin.com/d72e67f5 |
| 05:06.23 | jnfuller | To me, DID is an assigned number |
| 05:06.43 | jnfuller | one coming from a provider, really it's the same as an assigned sip user |
| 05:07.01 | jnfuller | logically, not at a protocol level |
| 05:07.33 | [TK]D-Fender | lanning: INVITE sip:s@10.0.1.254 SIP/2.0 <- because you didn't tell them where to send to |
| 05:07.48 | florz | how about "telling the receiver of the signalling some information that allows it to distinguish different application level destination addresses"? |
| 05:08.07 | lanning | great, I was looking at the from/to fields, not the invite line... :) |
| 05:08.13 | jnfuller | right, which leads me back to sip is luser@host |
| 05:08.41 | florz | yeah? |
| 05:08.56 | jnfuller | and back to rfc 3261, which was the original ask |
| 05:08.56 | florz | You mean, like, in SIP there is no way to accomplish that? |
| 05:10.11 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83) |
| 05:10.25 | jnfuller | I'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 |
| 05:10.49 | jnfuller | functionally, but in sip there's no such thing as did, just uri |
| 05:10.59 | carrar | Direct Inward Dialing |
| 05:10.59 | jnfuller | are we arguing the same side? |
| 05:11.20 | [TK]D-Fender | LESS FILLING!!! |
| 05:11.26 | carrar | TASTE GREAT |
| 05:11.31 | [TK]D-Fender | LESS FILLING!!!!!!!! |
| 05:11.31 | florz | jnfuller: I guess what you are calling "DID" is commonly called a "telephone number" |
| 05:11.38 | jnfuller | or sip uri |
| 05:11.38 | carrar | TASTE GREATER!! |
| 05:11.40 | [TK]D-Fender | grabs his ClueBat (tm) |
| 05:11.54 | [TK]D-Fender | carrar: Might makes right, and I am very VERY right! :p |
| 05:11.57 | jnfuller | there's really no such thing as a telephone number in sip |
| 05:12.01 | florz | jnfuller: no, a SIP URI is called a SIP URI, usually |
| 05:12.20 | carrar | TK, You wanna call that a DID? |
| 05:12.23 | carrar | Lets call that DID |
| 05:12.34 | carrar | I'm gonna call everything a DID |
| 05:12.48 | carrar | FARKLOR |
| 05:12.52 | [TK]D-Fender | goes off to confuse Wikipedia users some more. |
| 05:13.11 | florz | jnfuller: BTW, jnfuller@blah.baz is syntactically not a SIP URI |
| 05:13.24 | jnfuller | yeah yeah, it was an abstraction |
| 05:13.42 | florz | jnfuller: 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-Fender | carrar: did you do what you did when you said you did get the did I did tell you you did need to get? |
| 05:15.03 | carrar | I did |
| 05:15.08 | florz | that 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/ |
| 05:15.37 | jnfuller | conceptually yes |
| 05:15.39 | florz | SIP is the implementation of did nt, then? |
| 05:15.41 | jnfuller | I agree |
| 05:20.48 | jnfuller | but from a sip protocol perspective the term DID is markedroid bullshit |
| 05:21.00 | florz | because? |
| 05:21.29 | jnfuller | because if you actually read rfc's there is only user@host |
| 05:21.58 | florz | .o() |
| 05:22.15 | florz | BTW, no, there is not, not even close |
| 05:22.38 | florz | but that's immaterial to this discussion |
| 05:23.51 | *** join/#asterisk sergey (n=sergey@sergey-home.iks.ru) |
| 05:29.29 | jnfuller | How 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) |
| 05:32.58 | florz | I think the discussion was about DIDs, not about "addressing To headers". |
| 05:33.10 | jnfuller | so what's the actual problem, lanning |
| 05:34.44 | jnfuller | lanning: quick SIP question: To: format is <user>@<server>/<did> correct? |
| 05:35.28 | lanning | all inbound calls are going to "s", I thought it was something in the To: field, but it is in the INVITE field. |
| 05:36.18 | lanning | somehow I need to get the provider to send me the actual DIDs |
| 05:37.01 | jnfuller | are the pastebins from asterisk debug? |
| 05:37.18 | lanning | ya. |
| 05:37.27 | jnfuller | can you get one from tcpdump? |
| 05:37.35 | lanning | core set verbose 10; sip set debug |
| 05:37.41 | jnfuller | or tshark or snort, whatever |
| 05:38.03 | lanning | ya, just a sec |
| 05:40.01 | jnfuller | s 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 |
| 05:40.44 | lanning | well the sip debug showed "INVITE sip:s@10.0.1.254 SIP/2.0" so they are sending me "s", not the DID. |
| 05:40.57 | jnfuller | no, that's triaged by asterisk |
| 05:41.05 | jnfuller | s is a default extension |
| 05:41.06 | lanning | oh |
| 05:41.25 | jnfuller | that's why I said you needed to look at tshark, tcpdump or snort |
| 05:41.29 | jnfuller | what os are you on |
| 05:42.11 | lanning | CENTOS |
| 05:42.11 | jnfuller | ok so tcpdump should be available |
| 05:42.16 | lanning | I'll have a dump in a sec |
| 05:42.47 | jnfuller | if you cap it and look at it in wireshark you should be able to see what they are actually sending in the invite |
| 05:43.57 | jnfuller | and then you just need to check your context extension matching to make sure their presentations are accounted for |
| 05:46.28 | lanning | it has the "s" extension... |
| 05:46.39 | jnfuller | from the provider? |
| 05:46.53 | lanning | ya. it's weird |
| 05:46.57 | jnfuller | ok that's fucked |
| 05:47.03 | jnfuller | put in a troubl |
| 05:47.31 | lanning | yup, that's next. |
| 05:47.59 | jnfuller | I assume you're getting the call supposed to be to 4086367703 |
| 05:48.24 | florz | userpart s in the request URI is probably perfectly correct behaviour, anything else probably would be badly broken |
| 05:48.35 | jnfuller | is this peered from another asterisk server? |
| 05:48.43 | lanning | ya. main number is 4086367700, but the test DID is 4086367703. |
| 05:49.08 | florz | lanning: have you looked at the To-header? |
| 05:49.16 | lanning | I don't know what they are running. |
| 05:49.47 | jnfuller | You're looking at tcpdump how? |
| 05:50.19 | lanning | tcpdump -i eth0 -s 1500 -w sip.cap host 192.168.22.212 and udp port 5060 |
| 05:50.29 | jnfuller | I mean, you're sure you're looking at the wire and not the asterisk chatter? |
| 05:50.29 | lanning | then pulled into wireshark. |
| 05:50.55 | jnfuller | because you will also catch the internal sip chatter for wireshark |
| 05:51.00 | lanning | packet came from the provider's IP |
| 05:51.47 | jnfuller | ok 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 |
| 05:52.05 | jnfuller | unless that is a valid username, which in this case is not the case |
| 05:52.52 | florz | jnfuller: could you please justify that statement with RFC references? |
| 05:53.16 | jnfuller | I already did, florz |
| 05:53.30 | jnfuller | cut the shit, I',m trying to be helpful |
| 05:53.34 | florz | jnfuller: where? sorry if I missed it ... |
| 05:54.58 | jnfuller | so anyway if the packet did come from the provider it is malformed |
| 05:55.13 | jnfuller | and you should put in a trouble |
| 05:55.34 | florz | jnfuller: am I understanding you correctly that you are stating that the packet is malformed because the userpart in the request URI is 's'? |
| 05:56.01 | jnfuller | No, I'm saying that if it' |
| 05:56.10 | lanning | florz, in this case the "s" is not what I am expecting. |
| 05:56.20 | florz | lanning: because? |
| 05:56.34 | lanning | I am expecting the actual phone number, so I can switch based on it. |
| 05:56.34 | jnfuller | it's supposed to be addressed to 4086367703 and is addressed to s then it is wrong |
| 05:56.55 | jnfuller | so hey this is the tier 1 for asterisk so stop being such an ass |
| 05:57.18 | florz | lanning: why are you expecting it to have 4086367703 as the request URI's userpart? |
| 05:57.21 | jnfuller | lanning is here to get help |
| 05:57.37 | lanning | because that is the number that I dialed |
| 05:57.48 | florz | lanning: through the PSTN? |
| 05:57.53 | lanning | yes |
| 05:58.09 | lanning | to the provider, which should be passing it to me |
| 05:58.35 | florz | lanning: 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 |
| 05:58.36 | lanning | cell phone -> PSTN -> ITSP -> Asterisk |
| 05:58.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 05:58.54 | kaldemar | lanning: do you register to your provider, and if so, how? |
| 05:58.56 | florz | lanning: I assume that you are registering to the provider (via SIP, I mean)? |
| 05:59.07 | lanning | yes |
| 05:59.22 | lanning | via user.conf |
| 05:59.34 | lanning | s/r/rs/ |
| 05:59.36 | florz | lanning: well, then you are telling the provider in the REGISTER request which SIP URI to send calls for you to |
| 06:00.14 | florz | lanning: and there, you are probably telling it to send calls to s@<yourmachine>, which the provider, abiding by the standards, does |
| 06:00.19 | jnfuller | does the context you register for that user have any sort of pattern match for this number? |
| 06:00.34 | lanning | yes |
| 06:00.44 | jnfuller | is there a wildcard match? |
| 06:00.50 | kaldemar | you should do it in sip.conf with "register => username:secret@host/callbackextension" |
| 06:00.55 | lanning | exten = _4086367701,1,Goto(default|522|1) |
| 06:00.56 | lanning | exten = _4086367703,1,Goto(default|510|1) |
| 06:00.56 | lanning | exten = _X.,1,Goto(default|500|1) |
| 06:01.04 | kaldemar | if you're lacking the callbackextension, the provider will send s. |
| 06:01.13 | florz | lanning: so, there is nothing else the provider can do in this case |
| 06:01.41 | florz | kaldemar: that doesn't help, as it doesn't allow different number to be distinguished |
| 06:01.55 | lanning | so, I have to register every DID I have? |
| 06:01.58 | jnfuller | the provider doesn't actually send s, but it sends unscreened |
| 06:02.03 | lanning | I have never needed to do that before |
| 06:02.17 | jnfuller | so unless you have defined uac's asterisk will triage them as unknown |
| 06:02.36 | florz | lanning: either that, or look at headers - which is why I asked you whether you had looked at the To: header ... |
| 06:03.00 | lanning | the To header has the correct sip uri, the INVITE does not. |
| 06:03.20 | *** join/#asterisk AlmightyOatmeal (n=jamie@24-196-139-57.dhcp.mdsn.wi.charter.com) |
| 06:03.23 | florz | lanning: the INVITE does have the correct SIP URI, just not the one you are expecting |
| 06:03.43 | florz | lanning: it's the only one allowed there by the standard |
| 06:04.20 | lanning | so if I have 500 DID's I have to have 500 register lines? and register 500 times? |
| 06:04.46 | florz | lanning: no, you have to look at the headers, still |
| 06:04.58 | florz | lanning: or not use registration, if your provider offers that |
| 06:05.18 | lanning | how do I get asterisk to use the To header in extensions.conf? |
| 06:05.45 | florz | lanning: basically: not |
| 06:06.07 | florz | you'll have to do that yourself with your dialplan code |
| 06:06.11 | lanning | then do tell me I have to look at the To header. |
| 06:06.40 | jnfuller | you could wildcard match all calls and redirect a callerid num to a new context but that has the tendency to be buggy |
| 06:06.42 | lanning | I have never had this issue before, why just this provider? |
| 06:06.51 | florz | well, I am just telling you what you'll have to do from the point of view of the protocol |
| 06:07.02 | rue_more | I think I'm getting t the point where I can make my own co interfaces |
| 06:07.22 | rue_more | anyone want ulaw pots interfaces? |
| 06:07.25 | florz | lanning: maybe the other providers didn't implement SIP? |
| 06:07.49 | rue_more | heh use a scsi bus, haha |
| 06:08.03 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:08.06 | lanning | sip register worked just fine, and I got all the DIDs working, straight. |
| 06:08.21 | lanning | just this one is an issue. |
| 06:08.22 | florz | lanning: well, then you weren't usind SIP |
| 06:08.48 | lanning | well it was asterisk to a sip provider. that's all I can say. |
| 06:09.06 | jnfuller | do you have a peer set up in sip.conf to receive calls unregistered from the provider? |
| 06:09.41 | florz | lanning: well, no, obviously it wasn't a sip provider, had it been a sip provider, it hadn't worked |
| 06:09.50 | lanning | no just the users.conf with insecure = very |
| 06:10.03 | lanning | florz, go away if you are not helping |
| 06:10.35 | jnfuller | what revision of asterisk is this you are using |
| 06:10.37 | florz | lanning: oh, sorry, I am not helping by explaining the functioning of SIP? |
| 06:10.53 | lanning | 1.4.18.1 |
| 06:11.18 | jnfuller | set up a friend using the provider IP with insecure=port;invite |
| 06:11.29 | jnfuller | er port,invite |
| 06:11.31 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 06:11.48 | lanning | ok, just a sec |
| 06:11.52 | jnfuller | create it in an isolated contect |
| 06:11.55 | jnfuller | context |
| 06:12.35 | florz | lanning: and anyhow, I guess am capable of deciding when I want to go away all by myself, but thanks for your support ;-) |
| 06:13.39 | jnfuller | [peer] |
| 06:13.39 | jnfuller | <PROTECTED> |
| 06:13.40 | jnfuller | <PROTECTED> |
| 06:13.40 | jnfuller | <PROTECTED> |
| 06:13.40 | jnfuller | <PROTECTED> |
| 06:13.47 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 06:13.52 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.75) |
| 06:14.22 | jnfuller | and then try to match in the captive context while not registered |
| 06:14.34 | jnfuller | unless your provider forces registration |
| 06:15.33 | florz | jnfuller: you aren't seriously trying to tell lanning to not use registrations, are you? |
| 06:15.58 | jnfuller | depends on the provider |
| 06:16.15 | florz | well, yeah, of course it depends on the provider |
| 06:16.28 | florz | but 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) |
| 06:17.17 | sgupta` | hello |
| 06:17.28 | jnfuller | the definition still works if he is registered, right? |
| 06:17.44 | lanning | ok, never mind, I will have to call the provider in the morning. |
| 06:17.45 | sgupta` | any indian users here on the channel |
| 06:19.54 | florz | jnfuller: depends on what you mean by "works", and on what you are trying to accomplish, I guess |
| 06:20.46 | jnfuller | are you nine? |
| 06:21.22 | jnfuller | Seven, maybe? |
| 06:22.07 | florz | nope, I am just one person |
| 06:22.14 | jnfuller | so anyway insecure=very is deprecated |
| 06:22.30 | jnfuller | as in a seven-year-old, fucktard |
| 06:22.41 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 06:23.14 | lanning | ok, I will steer away from "very" |
| 06:23.15 | florz | You mean, like, I don't use such words as you are using yet, so I must be underdeveloped somehow? I see. |
| 06:23.27 | jnfuller | you've done nothing but pick apart anything anyone says in trying to help |
| 06:23.32 | jnfuller | very immature |
| 06:23.43 | *** join/#asterisk oej (n=olle@212.17.152.150) |
| 06:24.58 | jnfuller | if this is the sort of tier 1 asterisk users can expect then no wonder people go with commercial products |
| 06:25.22 | rue_mohr | actually |
| 06:25.24 | florz | You mean, like, it would be more mature to let you do your guessing completely void of any understanding of the underlying technology? |
| 06:25.27 | rue_mohr | if you buy digium hardware |
| 06:25.38 | rue_mohr | you get AWESOME support from digium |
| 06:26.03 | jnfuller | a lot of what you've said here is completely incorrect |
| 06:26.13 | florz | Like, instead of telling you that you are totally at the dead end, and even what the solution is? |
| 06:26.17 | rue_mohr | jnfuller, 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) |
| 06:26.34 | jnfuller | ok smartnick what is the solution |
| 06:26.41 | florz | jnfuller: well, could you point me to one incorrect thing I said? |
| 06:27.14 | jnfuller | well for one, the s in the invite is complete fallacy |
| 06:27.21 | j_kroon | hi 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. |
| 06:27.24 | jnfuller | find me the rfc |
| 06:27.48 | rue_mohr | jnfuller, you have problem? I dont see it back there |
| 06:28.31 | rue_mohr | jnfuller, there are a few people here who are real sharp, dont get me wrong, but there is a lot of background static |
| 06:28.34 | jnfuller | this is supposed to be a community of people helping each other, and the discussion list usually is but this is appalling |
| 06:28.50 | rue_mohr | j_kroon, cant help, sorry |
| 06:28.53 | florz | jnfuller: 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. |
| 06:29.15 | florz | sorry, s/x/s/ |
| 06:29.23 | jnfuller | so that's triaged by asterisk and defined by asterisk as a dialogue between the provider |
| 06:29.43 | jnfuller | it's not part of the protocol, it's a defined state by a sip dialogue |
| 06:32.22 | jnfuller | basically, we've been arguing the same point and you've been nitpicking like an idiot |
| 06:32.38 | florz | jnfuller: 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) |
| 06:33.40 | florz | jnfuller: 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 |
| 06:33.56 | jnfuller | exactly, but that doesn't mean the provider sends s if you don't define your location, which is what you alluded to |
| 06:34.15 | dshap | hey 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. |
| 06:34.26 | jnfuller | the provider sends to the default dialogue |
| 06:34.49 | florz | jnfuller: 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. |
| 06:34.51 | jnfuller | so we're all capable of being wrong or not 100% on our protocols so back off |
| 06:35.25 | jnfuller | you can register without providing a callback |
| 06:35.36 | jnfuller | which is what causes this issue |
| 06:35.44 | florz | jnfuller: what do you mean by "callback"? |
| 06:36.04 | *** join/#asterisk lmsteffan (n=laurent@114.69.191.239) |
| 06:36.04 | florz | jnfuller: registration uses so-called "contact URIs", if you mean those? |
| 06:36.15 | jnfuller | member:kaldemar |
| 06:36.15 | jnfuller | : |
| 06:36.15 | jnfuller | you should do it in sip.conf with "register => username:secret@host/callbackextension" |
| 06:36.25 | jnfuller | we discussed this earlier |
| 06:36.38 | jnfuller | sorry for the bad cut and paste there |
| 06:37.21 | florz | jnfuller: 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 |
| 06:37.27 | jnfuller | anyway 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) |
| 06:38.18 | florz | jnfuller: and in this case it was all about asterisk explicitly registering sip:s@<hismachine> as its contact URI ... |
| 06:39.21 | lanning | no it wasn't |
| 06:39.23 | jnfuller | yeah 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 |
| 06:40.02 | jnfuller | and after all this lanning still didn't get any relief |
| 06:40.19 | jnfuller | I'm sure florz can take it from here, lanning. He's the expert |
| 06:40.31 | lanning | asterisk is not registering s@host. it was registering 4086367700@host, and when 4086367703 was dialed I got "s" |
| 06:40.41 | florz | lanning: so, your asterisk sent a REGISTER for a different URI than you got the INVITEs to? |
| 06:40.50 | lanning | yes |
| 06:41.08 | florz | lanning: can you paste the REGISTER message somewhere? |
| 06:41.18 | jnfuller | my apologies to everyone not involved in all this stupid discussion |
| 06:41.25 | lanning | not now, as I am going to bed. |
| 06:41.35 | dshap | hah |
| 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) |
| 06:41.50 | florz | well, that makes helping rather difficult =:-) |
| 06:42.48 | florz | jnfuller: 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? |
| 06:43.10 | jnfuller | perhaps some tact may be in order in the future |
| 06:43.11 | sgupta` | I have problems with hangup and answer detection on trunk line, are there any Indian users here? |
| 06:43.23 | dshap | let 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? |
| 06:43.31 | dshap | from my * box |
| 06:44.21 | jnfuller | and I'm sorry I called you a fucktard |
| 06:44.32 | jnfuller | but you were being annoying |
| 06:45.33 | jnfuller | you mean the actual dtmf tones? |
| 06:46.05 | jnfuller | you need to check your inband dtmf settings |
| 06:46.20 | dshap | there's this web-based control panel for my SIP provider |
| 06:46.42 | dshap | under DTMF mode they have auto, rfc2833, inband, and info |
| 06:46.55 | dshap | i've tried enabling each of them and i still can't hear the tones |
| 06:47.15 | dshap | which leads me to believe there is something wrong with my server settings |
| 06:47.21 | dshap | i tried dtfmode=auto in sip.conf |
| 06:47.26 | dshap | as well as dtfmode=rfc2833 |
| 06:47.37 | jnfuller | what about your client? |
| 06:47.49 | dshap | i'm calling a cell phone |
| 06:47.57 | jnfuller | no, where you call from |
| 06:48.06 | dshap | that's my asterisk server |
| 06:48.07 | jnfuller | the sip phone, ata, whatever |
| 06:48.13 | dshap | oh |
| 06:48.16 | dshap | i'm using a .call file |
| 06:48.20 | dshap | that goes to an extension |
| 06:48.32 | dshap | the ultimate plan is to have it dial another system (not mine) and interact with a voice menu |
| 06:48.42 | dshap | by sending the appropriate DTMF tones at the right time |
| 06:48.44 | dshap | if that makes sense |
| 06:48.50 | dshap | right now it's not sending any tones |
| 06:48.55 | jnfuller | for an ivr you want to send inband |
| 06:49.05 | dshap | ~ivr |
| 06:49.06 | infobot | somebody said ivr was Interactive Voice Response |
| 06:49.19 | dshap | okay so how do i set that up |
| 06:49.53 | dshap | jnfuller: like if i was going to have my asterisk server dial my cell phone voicemail provider and type in my password |
| 06:49.57 | jnfuller | dtmf inband on both sides, i think, if you're using a call file |
| 06:50.17 | florz | jnfuller: 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 ... |
| 06:50.34 | dshap | by 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? |
| 06:50.39 | dshap | nothing else, right? |
| 06:51.05 | jnfuller | I've actually read so many rfc's I have trouble keeping all the info straight, florz |
| 06:51.45 | jnfuller | but I do work with an engineer who has never read them and keeps talking about dead cisco drafts as gospel, so I can relate |
| 06:51.57 | jnfuller | plus, I've had about twelve beers |
| 06:52.04 | florz | =:-) |
| 06:52.30 | jnfuller | so forgive me if I'm a bit stumbly |
| 06:52.39 | Qwell | jnfuller: Come back tomorrow then. |
| 06:52.40 | jnfuller | I would assume so, dshap |
| 06:53.02 | jnfuller | I've never tried to outpulse dtmf with a callfile |
| 06:53.12 | jnfuller | but the ivr will need to hear inband |
| 06:53.21 | jnfuller | otherwise it can't react on the tones |
| 06:53.39 | jnfuller | er, use a call file to call a context that outpulses dtmf |
| 06:53.51 | jnfuller | I know a callfile can't outpulse dtmf ;) |
| 06:54.41 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-82-2.hag.east.verizon.net) |
| 06:55.14 | dshap | jnfuller: i hear absolutely nothing. and in my CLI output it says that it is executing SendDTMF(1234) and reports no issues with that |
| 06:56.36 | dshap | could this have something to do with the audio codec configuration? |
| 06:57.15 | jnfuller | can you talk over the channel when you don't use a callfile? |
| 06:58.07 | dshap | i 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 |
| 06:58.15 | dshap | and hear the person on the other PSTN line |
| 06:58.25 | dshap | so inbound and outbound audio from talking on the channel work fine |
| 06:58.30 | dshap | as does audio from Playback() |
| 06:58.37 | jnfuller | and can you pass dtmf when not using a callfile? |
| 06:58.57 | dshap | i haven't tried |
| 06:58.59 | dshap | i'll try right now |
| 06:59.16 | dshap | im going to make it so that i call my server, press extension 1, and then my server should execute a SendDTMF |
| 06:59.17 | dshap | is that okay? |
| 06:59.47 | jnfuller | I suppose that would work |
| 07:00.13 | shido6 | or RTP being blocked |
| 07:00.16 | jnfuller | but I was more thinking inband all the way like hairpinning an incoming call |
| 07:00.44 | dshap | not sure what you mean by that |
| 07:00.53 | jnfuller | so you can send the dtmf from your cell to the pstn and check and make sure the dtmf isn't being turfed |
| 07:01.03 | dshap | ohhh |
| 07:01.08 | dshap | i can check that also |
| 07:01.12 | dshap | so like |
| 07:01.19 | dshap | my asterisk server sets up a bridge between 2 PSTN lines |
| 07:01.28 | dshap | and i press numbers on one of the phones |
| 07:01.29 | shido6 | or dtmfmode |
| 07:01.35 | dshap | and see if ic an hear on the other side |
| 07:02.19 | jnfuller | right |
| 07:03.16 | jnfuller | once 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) |
| 07:03.34 | dshap | ok we've got issues |
| 07:04.02 | dshap | i made the changes you suggested earlier |
| 07:04.11 | dshap | dtmfmode=inband and inband on my provider control panel |
| 07:04.21 | dshap | and now i can't even dial extensions |
| 07:04.26 | dshap | my server isn't receiving DTMF anymore |
| 07:04.40 | jnfuller | ok so switch them back and make the change in the extension to inband post-dial |
| 07:04.58 | dshap | k 1 sec |
| 07:04.59 | shido6 | inband and ulaw work well , rfc2833 and everything else should work |
| 07:05.50 | dshap | the issue is definitely the INBAND setting on my provider's web contorl panel |
| 07:06.58 | shido6 | if its on a ctrl panel there may be some lag between your update and the propogation across their network |
| 07:07.12 | jnfuller | so establish the call using what actually works for your provider and then try forcing SIPDTMFMode(inband) in the context you use for the callfile |
| 07:08.01 | jnfuller | check the syntax for that, that is off the top of my head. Corrections, anyone? |
| 07:08.16 | dshap | okay sorry i'm very new to all of this and you guys are a little ahead of me right now |
| 07:08.23 | dshap | i really appreciate the help and i'm going to try everything you say |
| 07:08.37 | dshap | i just verified the SendDTMF is not working even when i use it without a call file |
| 07:08.51 | dshap | now im gonna check the PSTN --> * --> PSTN thing |
| 07:09.24 | jnfuller | do you know the difference between inband and rfc2833? |
| 07:09.30 | jnfuller | functionally, I mean? |
| 07:10.01 | dshap | ok |
| 07:10.04 | dshap | no, i don't |
| 07:10.08 | dshap | but i just made an important discovery |
| 07:10.14 | dshap | her'es what i did |
| 07:10.19 | dshap | i have 2 PSTN lines, A & B |
| 07:10.27 | jnfuller | inband is in the audio path of the phone conversation as audio |
| 07:10.29 | dshap | i used A to call my asterisk box |
| 07:10.41 | dshap | [ok, gotcha] |
| 07:10.51 | dshap | then in the extension, i used Dial() to call B |
| 07:11.07 | dshap | which bridged the 2 channels |
| 07:11.14 | dshap | so they could talk to each other through my server |
| 07:11.16 | jnfuller | info and rfc2833 are signalling methods that use the rtp path to send a representation of the digits and not the actual audio |
| 07:11.54 | dshap | [alright so audio SHOULD work because i can talk on the channel] |
| 07:11.59 | dshap | here's the thing |
| 07:12.05 | jnfuller | well there's the gotcha |
| 07:12.12 | dshap | when i pressed a button on phone B, i could hear the tone on phone A |
| 07:12.14 | dshap | BUT |
| 07:12.15 | jnfuller | pstn sets have their own dtmf tone generators |
| 07:12.18 | dshap | when i pressed a button on phone A |
| 07:12.26 | dshap | all i heard was a little blip/click on phone B |
| 07:12.37 | jnfuller | digital set? |
| 07:12.38 | dshap | it was being cut off or something |
| 07:12.53 | dshap | one is a cell phone, the other is a landline |
| 07:13.02 | dshap | A = landline, B = cell phone |
| 07:13.05 | dshap | but listen |
| 07:13.09 | dshap | i tried to make the same call |
| 07:13.12 | dshap | withotu my asterisk server |
| 07:13.17 | dshap | i just picked up my landline and called my cell phone |
| 07:13.22 | dshap | and then the tones worked in both directions |
| 07:13.31 | *** join/#asterisk stijnbe (n=stijnbe@router.begen1.office.netnoc.eu) |
| 07:13.32 | dshap | my server or my provider is fucking it up |
| 07:13.56 | *** join/#asterisk stijnbe (n=stijnbe@router.begen1.office.netnoc.eu) |
| 07:14.20 | jnfuller | could be, some cells have fixed tone durations |
| 07:15.00 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:15.07 | dshap | you think it's working and the issue is that i'm using a cell phone? |
| 07:15.32 | jnfuller | I think that there are a lot of variables to go wrong |
| 07:15.40 | dshap | ok i think i've gotten that out though |
| 07:15.46 | dshap | i called my server with my landline |
| 07:15.52 | dshap | and did SendDTMF |
| 07:15.53 | dshap | heard nothing |
| 07:16.29 | jnfuller | so you call and the context answers, does a SendDTMF(1) and you hear no audio |
| 07:16.48 | dshap | yep |
| 07:16.50 | dshap | that is correct |
| 07:17.01 | dshap | SendDTMF(1234) actually |
| 07:17.02 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:17.03 | florz | jnfuller: no, INFO uses the signalling path, which is exactly what sets it apart from RFC2833 |
| 07:17.08 | jnfuller | try adding a SendDTMFMode(inband) before that |
| 07:17.19 | jnfuller | yeah that's right, info isn't in rtp |
| 07:17.23 | dshap | okay so the priority immediately before the SendDTMF |
| 07:17.31 | jnfuller | yep |
| 07:17.35 | dshap | trying that right now |
| 07:17.44 | jnfuller | anyone know what the actual syntax is for that |
| 07:18.53 | dshap | jnfuller: [Jun 3 00:18:26] WARNING[16811]: pbx.c:1832 pbx_extension_helper: No application 'SendDTMFMode' for extension (intro_menu, 3, 2) |
| 07:19.10 | jnfuller | I think it's setdtmfmode |
| 07:19.26 | jnfuller | just a sec |
| 07:19.35 | dshap | i dont see it on the application list on voip-info |
| 07:19.43 | dshap | unless it's a channel variable |
| 07:20.52 | jnfuller | just gonna log into a server |
| 07:21.43 | dshap | k |
| 07:21.50 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
| 07:21.52 | jnfuller | SIPDtmfMode(inband|info|rfc2833) |
| 07:22.07 | dshap | okay so SIPDtmfMode(inband) then? |
| 07:22.29 | Milad | is any way to manipulate gotoiftime in other date like jalali etc ... ? |
| 07:22.33 | dshap | and 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) |
| 07:22.51 | jnfuller | bleorgh,n,Set(SipDTMFMode=inband) ??? |
| 07:23.05 | jnfuller | like I said, I am fuzzy on the syntax for that command |
| 07:23.31 | dshap | according to voip-info it's an application and not a variable |
| 07:23.37 | dshap | so it'd be SIPDtmfMode(inband) i believe |
| 07:23.38 | dshap | ill try it |
| 07:23.57 | AlmightyOatmeal | does one really need a fxo/fsx channels if one intends to use wifi/voip phones and softphones? |
| 07:24.37 | jnfuller | http://www.the-asterisk-book.com/unstable/applikationen-sipdtmfmode.html |
| 07:24.39 | dshap | jnfuller: IT WORKED |
| 07:24.44 | jnfuller | yeah that's the right syntax |
| 07:24.47 | jnfuller | cool |
| 07:24.49 | dshap | i heard the tones! |
| 07:24.53 | dshap | do you think this will work for call files |
| 07:24.56 | dshap | and everything else? |
| 07:25.04 | dshap | should i just leave dtmfmode=inband in my sip.conf? |
| 07:25.12 | dshap | or do i always have to set the dtmf mode right before i send dtmf? |
| 07:25.21 | jnfuller | well didn't that break your other calls? |
| 07:25.36 | dshap | i'm pretty sure what broke the calls was the web control panel setting |
| 07:25.38 | jnfuller | ah |
| 07:25.41 | dshap | i dunno ill try |
| 07:25.48 | dshap | with web=auto and dtmfmode=inband in sip.conf |
| 07:26.23 | jnfuller | You 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 |
| 07:26.51 | dshap | yep |
| 07:26.52 | jnfuller | unless the ivr is sip and understands info or rfc2833 |
| 07:26.59 | dshap | doesn't work with dtmfmode=inband |
| 07:27.09 | dshap | gotcha |
| 07:27.15 | dshap | alright now im gonna try this with my voicemail thing |
| 07:27.24 | jnfuller | just like a person, if the box can't hear the tones it won't work |
| 07:27.33 | dshap | right |
| 07:27.38 | dshap | and if the box CAN hear the tones |
| 07:27.41 | dshap | hopefully it WILL work :) |
| 07:28.38 | dshap | that shouldn't mess up anything for that channel after i call it |
| 07:28.39 | dshap | right? |
| 07:28.52 | dshap | like it shouldn't change the way a caller can interact with my extensions by dialing numbers |
| 07:28.53 | dshap | right? |
| 07:29.21 | jnfuller | well if you do it in the context for outbound for the function you need the change won't be global |
| 07:29.36 | dshap | true |
| 07:29.38 | dshap | alright |
| 07:30.04 | dshap | another quick question |
| 07:30.16 | dshap | you know the tone you hear before you're supposed to leave a recording? |
| 07:30.22 | dshap | like, "Please leave a message after the beep" |
| 07:30.24 | dshap | and then you hear the beep |
| 07:30.26 | dshap | is that DTMF? |
| 07:30.59 | jnfuller | I don't think so, I don't know if there is even a standard for that tone |
| 07:31.25 | dshap | would it be possible to detect it on a channel with asterisk? |
| 07:32.18 | jnfuller | there is an answering machine detection app but I don't know if it is production reliable |
| 07:32.55 | dshap | ah there's some stuff on google i found |
| 07:32.59 | dshap | ill check it out thanks for the tip |
| 07:33.44 | *** join/#asterisk acehunky (n=acehunky@123.252.144.92) |
| 07:33.58 | jnfuller | no prob, glad you got it working |
| 07:34.09 | jnfuller | the command is AMD |
| 07:37.32 | dshap | hmm |
| 07:37.50 | dshap | the D option on the Dial() app doesn't seem to work the same way |
| 07:38.09 | dshap | i tried setting SIPDtmfMode(inband) right before a Dial(SIP/number@trunk,20,D(1234)) |
| 07:38.29 | dshap | which is supposed to send the DTMF before the channels are bridged |
| 07:38.37 | dshap | any ideas? |
| 07:41.25 | jnfuller | perhaps D doesn't respect the settings in the context |
| 07:43.20 | jnfuller | you 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? |
| 07:43.39 | jnfuller | ugly hack |
| 07:43.43 | dshap | truth |
| 07:43.54 | dshap | i might not need D |
| 07:43.56 | dshap | just thought i'd ask |
| 07:44.15 | dshap | to be honest, i really just want to program my asterisk box to leave me a voicemail without calling my phone right now |
| 07:44.21 | dshap | by calling my AT&T voicemail backdoor number |
| 07:44.39 | dshap | i took out a stopwatch and timed how long the wait should be before it starts playback but it doesn't seem to be working |
| 07:45.51 | jnfuller | hmmm why not send yourself an sms instead? |
| 07:46.24 | dshap | hah well the reason i want to learn how to do this is a long and complicated story |
| 07:46.34 | jnfuller | the best kind |
| 07:46.40 | dshap | also i have no idea how to send SMS from my asterisk server |
| 07:46.44 | dshap | im sure i would need a special provider for that |
| 07:46.45 | dshap | which i dont have |
| 07:46.57 | jnfuller | http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 07:47.16 | dshap | isn't that expensive? |
| 07:47.30 | jnfuller | depends on your provider, I get free inbound |
| 07:47.40 | dshap | how much do they charge u |
| 07:47.43 | dshap | oh |
| 07:47.45 | dshap | free inbound |
| 07:47.45 | dshap | i c |
| 07:47.48 | dshap | how much for outbound |
| 07:48.23 | jnfuller | I think my plan is something like 200 messages and then 25 cents after that |
| 07:48.41 | dshap | for how much $ if you dont mind me asking |
| 07:48.46 | jnfuller | but the ones I send myself through an sms gateway are to the phone so I don't get charged |
| 07:48.47 | dshap | that's per month i'm assuming right? |
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| 07:49.30 | jnfuller | I think the equivalent non-discounted version of what I have is around $25 canadian a month |
| 07:49.37 | jnfuller | i work for a provider |
| 07:49.46 | dshap | wait you're talking about your server sending & receiving SMS |
| 07:49.48 | dshap | not your mobile phone |
| 07:49.49 | dshap | right? |
| 07:50.03 | jnfuller | no I was talking about my plan |
| 07:50.25 | dshap | if i have an AT&T wireless plan with unlimited text messaging |
| 07:50.30 | dshap | can i somehow get my asterisk server to send SMS using that? |
| 07:51.19 | dshap | i literally know nothing about SMS |
| 07:51.34 | jnfuller | if you want to send out from the phone you will need to use chan_mobile |
| 07:52.26 | dshap | ah i c (just looked that up) |
| 07:52.33 | dshap | yea i think that's an adventure for another day |
| 07:52.34 | dshap | hah |
| 07:53.01 | dshap | is it possible that the default SendDTMF between-digit delay is too small for my AT&T voicemail server to be able to understand? |
| 07:53.15 | jnfuller | I went and looked on our server and I'm actually using jabber im and not sms |
| 07:53.39 | dshap | when i call my voicemail backdoor, the first thing i hear is "please enter the mailbox number and press #" |
| 07:54.28 | jnfuller | try increasing the tone duration |
| 07:56.17 | dshap | okay wtf |
| 07:56.25 | dshap | i went back to my extension that uses the Dial() app |
| 07:56.32 | dshap | PSTN --> * --> another PSTN |
| 07:56.38 | dshap | and for the other PSTN i put my voicemail backdoor number |
| 07:56.58 | dshap | and when i do that setup, i can't interact with the IVR through the PSTN phone i'm calling on |
| 07:58.13 | dshap | actually |
| 07:58.16 | dshap | that problem was present all along |
| 07:58.17 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:58.31 | dshap | with my 2 phones that i have |
| 07:58.51 | jnfuller | hmmm swap the two phones and if the problem moves beat one to death with a hammer |
| 08:00.30 | dshap | i tried each phone |
| 08:00.51 | dshap | can'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) |
| 08:01.18 | dshap | i have a feeling my call file SendDTMF doesnt work either (checking that now) |
| 08:01.24 | jnfuller | what are you using for fxs/fxo |
| 08:01.36 | dshap | im not using those |
| 08:01.40 | dshap | it's just an IP connection |
| 08:01.42 | dshap | on the server |
| 08:01.56 | jnfuller | so this is all voip |
| 08:02.43 | dshap | yes |
| 08:02.56 | dshap | as i thought! |
| 08:03.03 | dshap | even with SIPDtmfMode(inband) |
| 08:03.09 | dshap | when i initiate a call with a call file |
| 08:03.25 | dshap | and hook it to an extension that does SIPDtmfMode(inband) and then SendDTMF() |
| 08:03.27 | dshap | it doesnt work |
| 08:04.02 | dshap | (my server is all VoIP....i'm trying to interact with PSTN lines via my SIP trunk provider) |
| 08:05.02 | dshap | so 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 |
| 08:05.04 | dshap | however |
| 08:05.09 | dshap | if my server calls ME |
| 08:05.12 | dshap | the same commands fail |
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| 08:13.49 | dshap | drmessano: you there? |
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| 08:48.04 | WeazelON | hey guys, i was wondering if anyone knows, how do i change the vm-intro language on a specific extention |
| 08:48.14 | WeazelON | extension* |
| 08:48.37 | WeazelON | i have 2 kinds of language prompts already in the PBX |
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| 08:53.06 | WeazelON | anyone ? |
| 08:56.00 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 08:56.05 | WeazelON | :( |
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| 09:05.37 | jgoo | I have *8 working |
| 09:05.55 | jgoo | however there is a delay, about 1.5 rings before it picks up - can I reduce this delay? |
| 09:06.09 | jgoo | s/\\r\\n/, |
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| 11:05.28 | CRCinAU | can anyone help me with what's going on here? |
| 11:05.29 | CRCinAU | http://crc.pastebin.com/d1d2d2d5f |
| 11:05.44 | CRCinAU | I'm not sure why I get a broken pipe :\ |
| 11:10.44 | *** part/#asterisk CRCinAU (n=CRCinAU@203.56.246.91.crc.id.au) |
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| 11:19.07 | AlmightyOatmeal | grrr, i'm getting a registration timeout for BroadVoice :( |
| 11:21.02 | devyll | any 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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| 11:38.21 | devyll | I guess the only way is to replace that audio file with my audio file ?! |
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| 12:14.25 | shay | anyone here using SIP wants to try giving me a call? |
| 12:14.33 | shay | or is there a sip test that calls you? |
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| 12:31.55 | xa0z | Is it possible to have pfsense, and asterisk running on the same box, without problems? |
| 12:32.41 | [TK]D-Fender | xa0z: As long as pfsense doesn't mess with the ports * needs. |
| 12:34.07 | xa0z | Okay, I'm going to assume no one has really discussed pfsense, and asterisk? |
| 12:34.39 | *** join/#asterisk ms110 (n=Shehzad@122.169.79.177) |
| 12:34.50 | xa0z | I'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. |
| 12:34.50 | ms110 | hello |
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| 12:35.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:36.03 | Great_Anta_Baka | in my cdr records my src field is blank. How do I set that? |
| 12:36.43 | [TK]D-Fender | xa0z: as long as packets get in & out of * unmolested you'll be fine |
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| 12:39.28 | xa0z | Reading on pfsense's site, looks like it would be more difficult to set it up on one box. |
| 12:39.33 | juanIMP | good morning :D |
| 12:41.02 | [TK]D-Fender | xa0z: 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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| 12:42.19 | xa0z | Ok. |
| 12:43.09 | [TK]D-Fender | damn a lot fo caps failure today... |
| 12:46.00 | xa0z | What would be a good system to use for asterisk? Something like a ITX, multiple lan, etc. |
| 12:47.12 | xa0z | I have an HP T5700 1.0Ghz 256mb ram, 256mb flash, 1 ethernet, 4 usb (and a usb ethernet adapter). Would that be suitable? |
| 12:52.14 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 12:53.40 | ariel_ | 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-Fender | xa0z: Depends what you're doing with * |
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| 13:18.18 | wwalker | where can I find a list of what the Reason: codes in an Originate response mean? |
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| 13:25.36 | wwalker | frame.h appears to be the place |
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| 13:34.42 | ThatKidKel | I'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" |
| 13:34.47 | ThatKidKel | any suggestions please? |
| 13:35.27 | [TK]D-Fender | ThatKidKel: qualify=yes <- |
| 13:35.53 | ThatKidKel | [TK]D-Fender.. Already on |
| 13:36.55 | [TK]D-Fender | ThatKidKel: You'd have to share actual configs and SIP debug.... |
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| 13:39.04 | rue_mohr | isn't there a keepalive thing in there on some stuff? |
| 13:39.15 | ThatKidKel | i fond the issue |
| 13:39.19 | ThatKidKel | customer swore that the firewall was off |
| 13:39.26 | ThatKidKel | firewall was on, but the sip alg was off |
| 13:39.35 | rue_mohr | :) |
| 13:39.53 | ThatKidKel | they refused to let me into the firewall.. finally i convinced them |
| 13:39.55 | ThatKidKel | i hate people |
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| 13:42.49 | ajohnson | has fired customers over such issues :) |
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| 14:00.24 | jplank | can anyone comment on Rhino's hardware? |
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| 14:03.28 | [TK]D-Fender | jplank: channel banks are pretty decent, wouldn't use anything else though |
| 14:03.42 | jplank | really? |
| 14:04.12 | jplank | I was looking at their Ceros servers, matched with their cards. Not a good route? |
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| 14:05.02 | jaytee | hello |
| 14:05.08 | jaytee | I am not dave |
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| 14:08.46 | ricko73 | jplank: check in #rhino |
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| 14:11.16 | [TK]D-Fender | jplank: Their cards seem very 3rd tier. Wouldn't touch them. |
| 14:11.35 | [TK]D-Fender | jplank: Might be helpful if you actually stated your needs |
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| 14:12.51 | beek | morning [TK]D-Fender jaytee ricko73 |
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| 14:13.05 | jaytee | mornin beek |
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| 14:16.15 | [TK]D-Fender | mornin' |
| 14:16.48 | jplank | fender - 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 |
| 14:17.01 | jplank | their Ceros boxes seemed perfect to my needs |
| 14:18.58 | [TK]D-Fender | jplank: So ... what.. the RAM doesn't work with your MB? |
| 14:19.00 | ariel_ | 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. |
| 14:19.40 | ariel_ | But this point of biz I only use there channel banks. |
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| 14:21.06 | jplank | fender - 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 |
| 14:21.36 | jplank | The manufacturer usually gets the issue resolved, but then I'll order the same config, and have the same issues |
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| 14:21.44 | jplank | its too much of a PITA to deal with |
| 14:22.10 | jplank | I'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 |
| 14:22.28 | jplank | most people in here are |
| 14:22.38 | jplank | I 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 ;) |
| 14:22.58 | ariel_ | 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-Fender | jplank: Dell, IBM, HP. Get a box you can get SERViCE on. |
| 14:25.09 | jplank | my 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 |
| 14:26.57 | DarthPointer | ariel_: have you had any performance issues w/ the software RAID & * |
| 14:27.53 | jplank | ariel_: these servers look nice, thanks |
| 14:27.58 | wackypl | DarthPointer: hello, i ask about button click2dial remember ? |
| 14:28.07 | ariel_ | 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 |
| 14:28.08 | DarthPointer | wakcypl, yes |
| 14:29.19 | DarthPointer | gotcha; 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? |
| 14:29.21 | wackypl | DarthPointer: you said to me about asteridex but |
| 14:29.24 | [TK]D-Fender | jplank: Poor plan |
| 14:29.38 | wackypl | this application is not talk with browser |
| 14:29.49 | ariel_ | 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. |
| 14:30.39 | wackypl | DarthPointer: I'm looking for something like this http://e0800.pl/clickone27012008/clickone.html |
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| 14:31.03 | DarthPointer | wackypl: what application are you looking to use your click to dial for? (checking) |
| 14:32.08 | wackypl | click to dial with can talk on a browser using a microphone and headphones |
| 14:32.25 | jplank | I wasn't trying to get past 5.2 actually |
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| 14:33.25 | jplank | it 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 |
| 14:33.25 | DarthPointer | wacky: sorry, is that IE only? Can't check it. So you want a web based softphone? |
| 14:33.27 | Baylink | In 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? |
| 14:33.47 | wackypl | DarthPointer: DarthPointer yes only IE |
| 14:33.56 | wackypl | DarthPointer: using java & activex |
| 14:34.16 | jplank | fender - whats a poor plan? |
| 14:34.26 | pheller | anyone 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 # ? |
| 14:36.16 | DarthPointer | wackypl: I haven't deployed anything like that, but here's a link: http://www.pernau.at/kd/voip/ActXPhone/ to an ActiveX SIP Softphone |
| 14:36.35 | ariel_ | that reminds me I need to ask Patrick what's up with the board testing he is doing for us.... |
| 14:36.37 | DarthPointer | wackypl: I would start with the list here: http://www.voip-info.org/wiki/view/VOIP+Phones, under soft phones though |
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| 14:37.52 | wackypl | DarthPointer: THX |
| 14:38.05 | wackypl | DarthPointer: I HAVE TO GONE, i write to you later |
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| 14:50.22 | wwalker | anyone 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-Fender | wwalker: WaitforSilence() |
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| 14:51.05 | wwalker | All the searches I do tell me about issues regarding asterisk voicemail, no asterisk calling someone else voicemail. |
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| 14:51.33 | wwalker | [TK]D-Fender: got any idea what an optimum wait time is? |
| 14:51.57 | [TK]D-Fender | wwalker: "till its done" |
| 14:53.31 | wwalker | problem there is some answering machine will hang up because we were silent too long |
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| 14:57.05 | DarthPointer | wwalker: this was jsut discussed last night on list; Answering Machine Detect- http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD |
| 14:57.18 | DarthPointer | walker: although there was some question as to weather or not it was enterprise grade |
| 14:59.08 | [TK]D-Fender | wwalker: WaitforSilence() <------- |
| 15:00.29 | wwalker | DarthPointer: thank you |
| 15:01.50 | wwalker | [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. |
| 15:03.08 | ricko73 | is there a way when building Asterisk to have it grab the sound files locally instead of downloading them every time? |
| 15:05.35 | wwalker | DarthPointer: which list? I don't see anything about answering machine detection in the archives for asterisk-user |
| 15:06.02 | kaldemar | ricko73: prevent the download by de-selecting sound packages in make menuselect |
| 15:06.11 | DarthPointer | wwalker; sorry, on IRC last night |
| 15:06.43 | ricko73 | kaldemar: I don't think you understand me. We need the sound files, but want the build process to grab them locally |
| 15:07.39 | ricko73 | if they already exist on my computer, why should I have to download them every time I compile a new version |
| 15:08.31 | kaldemar | do you already have the files in the right place before compilation? |
| 15:10.00 | kaldemar | if not, just copy them to the right directory. |
| 15:12.25 | ricko73 | kaldemar: what do you mean by "the right place"? In the target directory? |
| 15:13.02 | ricko73 | or 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-Fender | wwalker: AMD first, WaitForSilence() second |
| 15:14.15 | kaldemar | ricko73: i mean sounds directory by the right place, /var/lib/asterisk/sounds by default. |
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| 15:15.01 | spck | what do you guys think of virtualizing asterisk with virtualbox? |
| 15:16.27 | kaldemar | ricko73: you could try to put the sounds tarball into asterisk-x.x.x.x/sounds/ that's where it downloads them. |
| 15:16.27 | ariel_ | spck: just one thing comes to mind.... have a nice test lab.... |
| 15:16.48 | ricko73 | kaldemar: that's closer to what I thought |
| 15:16.51 | spck | ariel: but not as an actual deployment? |
| 15:17.02 | ricko73 | this is an automated build process for a cross-compiled system |
| 15:17.15 | ariel_ | spck: depends on what you are doing with the asterisk |
| 15:17.24 | ricko73 | that's why I'm trying to prevent downloading several megabytes every time we re-build |
| 15:17.50 | spck | basically i want to load balance between two physical boxes for small call center |
| 15:17.59 | ricko73 | it 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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| 15:24.51 | iratik | I 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 |
| 15:24.51 | iratik | <PROTECTED> |
| 15:26.50 | iratik | everything i'm thinking of seems awefully hackish though |
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| 15:31.02 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 15:31.05 | acxty | Hi guys |
| 15:31.19 | acxty | Is it possible that asterisk automatically makes a call |
| 15:31.33 | acxty | For example I want that it call me all the days at 5PM |
| 15:31.37 | acxty | can that be done? |
| 15:32.04 | russellb | Yes. |
| 15:32.13 | acxty | russellb, where can I find information on that |
| 15:33.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:33.51 | russellb | Asterisk 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. |
| 15:34.10 | russellb | You can do via a TCP socket (AMI / manager interface), or via dropping files into a directory (call files) |
| 15:34.27 | russellb | or the CLI ... but that's not intended for scripting. |
| 15:34.57 | acxty | russellb, thanks will search on that ;) |
| 15:35.12 | russellb | you're welcome. |
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| 15:37.16 | kaldemar | would be nice to have an optional call time parameter in call files. it would make timed operations much easier. |
| 15:37.31 | ricko73 | kaldemar: that did the trick. Thanks. Not sure why I didn't think of copying the files there before |
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| 15:42.36 | Zhad | kal> don't you do it by setting the timestamp of the .call file? |
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| 15:49.22 | [TK]D-Fender | Zhad: you do |
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| 16:03.56 | kannan | hello 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? |
| 16:04.37 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 16:04.38 | Kernel_Core | hi all |
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| 16:06.58 | Kernel_Core | I 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-Fender | kannan: How long? |
| 16:12.52 | Kernel_Core | and here is my pastebin for debug E1 |
| 16:12.53 | Kernel_Core | http://pastebin.ca/index.php |
| 16:13.00 | rue_mohr | so, 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-Fender | kannan: Don't forget the base dialplan parameters have a limt for sure themselves |
| 16:13.25 | [TK]D-Fender | rue_mohr: Generally, no. |
| 16:14.25 | rue_mohr | arg, so, in general how does it go? |
| 16:15.15 | [TK]D-Fender | rue_mohr: A SIP header is sent with the call to tell it to auto-answer on speaker. the end. |
| 16:15.38 | [TK]D-Fender | rue_mohr: *'s Page() is just a multi-callout + Meetme w/ 1-way audio |
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| 16:24.10 | rue_mohr | oh, so most of the hard stuff is already done |
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| 16:26.03 | timeshell_atwork | Happy Threesday!\ |
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| 16:29.40 | ctooley | timeshell_atwork, is that something like Three-Times-As-Much-Stuff-Will-Break-Today-Day? |
| 16:30.09 | timeshell_atwork | ctooley : No.... Threesday comes after Twosday? |
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| 16:33.40 | enzo | hi |
| 16:33.50 | rue_mohr | brb |
| 16:34.14 | enzo | I'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 ? |
| 16:34.21 | mort_gib | ctooley: I'll buy that explanation !! |
| 16:34.27 | enzo | i use asterisk-app-fax |
| 16:43.13 | coppice | enzo: you probably have multiple versions of spandsp on your machine |
| 16:43.28 | enzo | no only libspandsp1 |
| 16:43.48 | enzo | it's a fresh install of ubuntu jaunty in fact coppice |
| 16:44.26 | coppice | ubuntu may have installed a version of spandsp. I assume you installed a more recent one |
| 16:45.00 | enzo | no I haven't upgrade to a new libspan version |
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| 16:45.51 | enzo | i have libspandsp1 0.0.5-pre4-1 on y ubuntu (default version for ubuntu 9.04) |
| 16:46.30 | coppice | that's pretty ancient, but I think * is supposed to allow for that version OK |
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| 16:47.41 | enzo | well the asterisk-app-fax is version 2007-06-24, quite old also |
| 16:47.49 | kannan | [TK]D-fender, i am not able to find the docs that specify the limitastions for the base dialplans themselves |
| 16:47.57 | enzo | don't know why a fresh ubuntu gives so old packets |
| 16:48.14 | tzafrir_laptop | (trivia: libspandsp.so.3 of Debian packages is actually older than libspandsp.so.1 which is the same as upstream's SONAME) |
| 16:48.19 | kannan | [TK]D-fender, i am sure thats the case, as the values are very lengthy, sometimes whole sentences |
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| 16:48.45 | kannan | [TK]D-fender, i guess i will do it with DB tables, |
| 16:48.48 | tzafrir_laptop | enzo, and won't build with that specific version of spandsp, IIRC |
| 16:48.54 | rue_mohr | so now Ihave to see if voip-info has a paging page |
| 16:49.14 | tzafrir_laptop | A known and painful issue of the sid freeze. Just use app_fax :-( |
| 16:49.19 | enzo | tzafrir_laptop: maybe you have an idea for the fact that rxfax crashes ? |
| 16:49.39 | jeffspeff | i 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. |
| 16:49.39 | kannan | [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 |
| 16:49.40 | tzafrir_laptop | because it was built for a different version of spandsp? |
| 16:50.00 | rue_mohr | jeffspeff, you want it to act like a keyed system dont you |
| 16:50.16 | jeffspeff | if that's the term, then yes |
| 16:50.23 | DarthPointer | kernel_core; you did not paste the correct url |
| 16:50.24 | rue_mohr | jeffspeff, asterisk isn't a keyd system |
| 16:50.50 | rue_mohr | jeffspeff, DONT use polycom sets... that will save you a lot of hurt |
| 16:51.26 | rue_mohr | jeffspeff, there is something called sip2, which can try to emulate a keyed system, your luck may be limited |
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| 16:51.44 | jeffspeff | rue_mohr, i thought that * would have been able to do that |
| 16:52.22 | enzo | that's strange ubuntu made such an error tzafrir_laptop, it should have already been reported, and i find very few things on this |
| 16:52.23 | lasko | Is 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. |
| 16:53.04 | jeffspeff | rue_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? |
| 16:53.41 | tzafrir_laptop | enzo, I'm not really sure packages in Universe get QA |
| 16:54.29 | enzo | is there a way for me to know what version of wich soft I have to install tzafrir_laptop ? |
| 16:55.41 | enzo | in 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 |
| 16:56.26 | rue_mohr | jeffspeff, I just been thu this, I can tell you what I did, and all hte problems with it |
| 16:57.25 | [TK]D-Fender | kannan: 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-Fender | kannan: I don't know the hard number myself, but I'm sure you could generate an answer really easily |
| 16:57.51 | rue_mohr | jeffspeff, 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 |
| 16:58.26 | rue_mohr | jeffspeff, 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 |
| 16:58.50 | rue_mohr | jeffspeff, I know its not what you dont want to hear, but did you? |
| 16:58.53 | [TK]D-Fender | jeffspeff: What phone are you looking at? |
| 16:59.08 | jeffspeff | just one sec. on the phone with customer |
| 16:59.13 | rue_mohr | jeffspeff, nomatter how much tk tells you to get polycom phones, DONT DO IT |
| 16:59.43 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:00.50 | *** join/#asterisk xorl (n=xorl@sys-adm.org) |
| 17:01.21 | xorl | Hey guys, what's a good sip provider besides vitelity the service is anoying. |
| 17:01.22 | [TK]D-Fender | jeffspeff: 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. |
| 17:02.13 | jeffspeff | [TK]D-Fender, i was looking at the 4 line polycom phone |
| 17:02.52 | rue_mohr | a) its not a 4 line phone, it just has 4 line keys |
| 17:03.02 | rue_mohr | all polycom phones have too few buttons |
| 17:03.08 | rue_mohr | and they have severe dtmf issues |
| 17:03.21 | rue_mohr | and the documentationa and support sucks |
| 17:03.29 | [TK]D-Fender | rue_mohr: Only for you really. No-one else in here seems to share your issues |
| 17:03.29 | rue_mohr | what am I up to D)? |
| 17:03.49 | rue_mohr | [TK]D-Fender, you didn't do that goodle search for polycom dtmf asterisk |
| 17:03.51 | rue_mohr | I did |
| 17:03.53 | jeffspeff | [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-Fender | jeffspeff: What model? |
| 17:03.59 | rue_mohr | everyone has a problem with polycom dtmf |
| 17:04.35 | jeffspeff | [TK]D-Fender, soundpoint 440 |
| 17:04.38 | jeffspeff | *550 |
| 17:04.39 | Qwell | rue_mohr: name one other person |
| 17:04.45 | [TK]D-Fender | rue_mohr: really? Where are all these screaming frustrated users? not HERe, and you know they are the most advocated maker here. |
| 17:04.57 | rue_mohr | I dont want to post the 160000 google hits |
| 17:05.08 | rue_mohr | no, 1% of techs use irc |
| 17:05.15 | [TK]D-Fender | jeffspeff: IP 550 is very hard to recommend in their line-up. What kind of use will the actual user make of it? |
| 17:05.39 | jeffspeff | [TK]D-Fender, i think i read somewhere that you can register a seperate user for each "line" |
| 17:05.45 | rue_mohr | polycom phones also take up too much deskspace, and where the hell are my wall mount brackets!? |
| 17:06.05 | ariel_ | Loves and uses mainly Polycom phones.. And has no dtmf issues with them.... |
| 17:06.10 | rue_mohr | jeffspeff, in a month, I'll still help you get it working the way your asking for |
| 17:06.13 | [TK]D-Fender | rue_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. |
| 17:06.21 | Qwell | rue_mohr: http://www.888voipstore.com/polycom-wall-mount-kit-pr-18556.html |
| 17:06.34 | rue_mohr | I ordered some, its been 3 months |
| 17:06.45 | rue_mohr | from williams communication |
| 17:06.54 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 17:06.58 | [TK]D-Fender | jeffspeff: You looking to share a single phone amongst multiple users? |
| 17:07.12 | rue_mohr | hah no |
| 17:07.14 | [TK]D-Fender | rue_mohr: WILLIAMS is your problem then |
| 17:07.19 | rue_mohr | he's trying to emulate a keyed system |
| 17:07.27 | [TK]D-Fender | rue_mohr: Not yet. |
| 17:07.33 | jeffspeff | [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-Fender | jeffspeff: Ok, then throw the entire concept of "lines" out the door. |
| 17:07.59 | rue_mohr | your "4" line phone can take 12+ lines |
| 17:08.18 | rue_mohr | I'm sorry, I'll shut up, for now |
| 17:08.18 | jeffspeff | [TK]D-Fender, i understand that sip uses channels etc. |
| 17:08.30 | [TK]D-Fender | jeffspeff: 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 |
| 17:08.36 | rue_mohr | sits on the sidebench and watches this all unfold again |
| 17:08.45 | [TK]D-Fender | jeffspeff: thus its jsut down to how many CALLS you can juggle at a time. |
| 17:09.39 | [TK]D-Fender | rue_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-Fender | rue_mohr: We'll see the reaction one the new reality settles in. |
| 17:10.00 | rue_mohr | I bet he's gonna get a tdm400 card and have terrible echo problems |
| 17:10.35 | [TK]D-Fender | rue_mohr: Any more predictions to share with us Nostradumbass? ;) |
| 17:10.39 | rue_mohr | but I have to say, digiums support is AWESOME |
| 17:10.46 | kannan | [TK]D-fender, thanks again, i have still not got the actual limitations number, i will update when i do, |
| 17:10.52 | jeffspeff | [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-Fender | rue_mohr: Yes, you are having to work with them daily ;) |
| 17:11.20 | [TK]D-Fender | jeffspeff: When you call person A and they aren't there, isn't that where Vm kicks in? |
| 17:12.03 | jeffspeff | [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. |
| 17:12.36 | rue_mohr | jeffspeff, may I ask what kinda phone system they use now? |
| 17:12.50 | [TK]D-Fender | jeffspeff: * has call-pickup capabilities, but it isn't so simple as grabbing a "line button" |
| 17:12.58 | jeffspeff | rue_mohr, some shitty pstn |
| 17:13.16 | [TK]D-Fender | jeffspeff>rue_mohr, some shitty pstn <- huh? |
| 17:13.17 | mort_gib | I just upgraded asterisk from 1.4.24 to 1.4.25 |
| 17:13.38 | mort_gib | and 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?? |
| 17:13.51 | jeffspeff | [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-Fender | mort_gib: Plenty of nasty bugs in that rel, you may want to downgrade till .26 |
| 17:14.07 | mort_gib | :-( |
| 17:14.23 | mort_gib | Having probs with this install.... |
| 17:15.15 | mort_gib | Downgrade to 1.4.26 ?? BUt that's in rc is it not?? |
| 17:15.40 | rue_mohr | jeffspeff, do the phones say something like 'nortel' or 'meridian' on them? |
| 17:15.41 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:16.19 | [TK]D-Fender | mort_gib: ... downgrade to 1.4.24 UNTIL .26 gets released |
| 17:16.21 | rue_mohr | 'panasonic' ? |
| 17:16.35 | *** join/#asterisk oej (n=olle@212.17.152.150) |
| 17:16.38 | [TK]D-Fender | jeffspeff: Ok, that could be ANYTHING. |
| 17:16.48 | mort_gib | [TK]D-Fender: Ok, thanks! |
| 17:16.52 | jeffspeff | [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. |
| 17:17.22 | rue_mohr | holds his mouth |
| 17:17.24 | jeffspeff | and "line 2" user would roll to "line 3" user in same way. |
| 17:17.40 | jeffspeff | or am i way off in how i'm thinking about this? |
| 17:18.00 | [TK]D-Fender | jeffspeff: How many phones, how many lines? |
| 17:18.01 | rue_mohr | [TK]D-Fender, yea hows he doin? |
| 17:18.15 | [TK]D-Fender | rue_mohr: Still a little jumbled :) |
| 17:19.26 | jeffspeff | [TK]D-Fender, about 8 phones. # of lines doesn't matter because it's all voip/sip |
| 17:20.38 | [TK]D-Fender | jeffspeff: 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-Fender | jeffspeff: But forget the concept of "lines". There is no association of "line" toa call going to a phone. |
| 17:21.26 | jeffspeff | [TK]D-Fender, can you direct me to a link on how an ext like that would work? |
| 17:21.36 | jeffspeff | or the name of the feature? |
| 17:21.45 | [TK]D-Fender | jeffspeff: Go lookup "call pickup" on the WIKI |
| 17:21.47 | [TK]D-Fender | ~wikis |
| 17:21.47 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 17:22.12 | jeffspeff | [TK]D-Fender, ok, thanks for your help |
| 17:22.58 | rue_mohr | I dont think he really hear all that stuff I said |
| 17:23.39 | Qwell | rue_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? |
| 17:24.33 | rue_mohr | I'm interested to know if he takes the same path I did |
| 17:25.02 | [TK]D-Fender | rue_mohr: And I saw you prjecting your needs before he gained a better understanding of the natural call handling style of these phones with * |
| 17:25.28 | rue_mohr | to me, right off the bat he said he wanted a keyed system out of asterisk |
| 17:25.44 | rue_mohr | and he had looked some stuff up to know to say and ask what he did |
| 17:26.20 | jeffspeff | rue_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-Fender | rue_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-Fender | rue_mohr: So keep on pounding that square peg into the round hole :) |
| 17:27.09 | [TK]D-Fender | Qwell: What guy? |
| 17:27.14 | rue_mohr | and I made it fit! |
| 17:27.27 | rue_mohr | the peg and the hole both took a bit of a hit in the process |
| 17:27.38 | [TK]D-Fender | rue_mohr: congratulations.... how's that card of yours doing? |
| 17:27.53 | xorl | hmm, 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? |
| 17:27.57 | rue_mohr | well, since I called digium support and talked to them its a lot better |
| 17:28.26 | [TK]D-Fender | rue_mohr: How "passable" is it now? |
| 17:29.20 | rob0 | Hmmm, vitelity is working here. What quality issues? |
| 17:29.39 | rob0 | Oh, for DID I just use ipkall :) |
| 17:29.56 | rue_mohr | on friday I'm gonna apply a patch that should resolve most of the audio issues |
| 17:30.25 | xorl | rob0: 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) |
| 17:31.33 | Nilzao | hi guys |
| 17:32.31 | rob0 | echo, 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) |
| 17:33.02 | Nilzao | I can't find if * is able to host E1 port |
| 17:33.07 | Nilzao | is that possible? |
| 17:33.12 | [TK]D-Fender | rue_mohr: "should". So not yet. How far off are things? |
| 17:33.28 | [TK]D-Fender | Nilzao: It can't. Asterisk is SOFTWARE |
| 17:33.49 | [TK]D-Fender | Nilzao: However Asterisk supports all sorts of HARDWARE that will let you interface with an E1 |
| 17:34.08 | Nilzao | [TK]D-Fender: ok then i can tell the interface E1, that i'm hosting lines? |
| 17:34.23 | jeffspeff | xorl, have you looked at Teliax? http://teliax.com |
| 17:34.29 | [TK]D-Fender | Nilzao: what does "hostling lines" mean? |
| 17:34.42 | rue_mohr | well, some of the audio is a little funny, BUT they resolved the muting issue that none of you could help with |
| 17:34.52 | rue_mohr | seems that there isa setting in the echo canceler |
| 17:35.22 | Nilzao | [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 |
| 17:36.04 | rue_mohr | <PROTECTED> |
| 17:36.06 | [TK]D-Fender | Nilzao: 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? |
| 17:36.24 | rue_mohr | its usually set to 24, 3 or 4 works great to stop it from muting during the conversation |
| 17:37.03 | [TK]D-Fender | rue_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 |
| 17:37.14 | Nilzao | [TK]D-Fender: have 2 E1 ports on * , and 1 E1 PBX |
| 17:37.41 | Nilzao | [TK]D-Fender: the first E1 port on * receive outside calls |
| 17:37.44 | [TK]D-Fender | Nilzao: 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-Fender | Nilzao: Ok, sounds like "yes" |
| 17:38.05 | Nilzao | [TK]D-Fender: yes |
| 17:38.15 | [TK]D-Fender | Nilzao: so * can act as CPE to the telco, and NET to your PBX |
| 17:38.17 | Nilzao | [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-Fender | Nilzao: Of course its possible |
| 17:38.47 | Nilzao | [TK]D-Fender: thanks, any keyword to google it? |
| 17:38.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:39.10 | [TK]D-Fender | Nilzao: What is there to goole? |
| 17:39.12 | [TK]D-Fender | Google* |
| 17:39.18 | ricko73 | I 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" |
| 17:39.27 | Nilzao | [TK]D-Fender: I can't thinkn how i ask it to google... |
| 17:39.37 | Nilzao | [TK]D-Fender: please google, how i host E1 ports |
| 17:39.41 | [TK]D-Fender | Nilzao: ask WHAT? |
| 17:39.47 | rob0 | um ... should we party with lemon jello? |
| 17:39.48 | Nilzao | [TK]D-Fender: or kinda something |
| 17:40.16 | [TK]D-Fender | Nilzao: 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. |
| 17:40.16 | Nilzao | [TK]D-Fender: i will try your way "* between telco and PBX" |
| 17:40.53 | [TK]D-Fender | Nilzao: Drop the word "host". It does not apply to this application at all. |
| 17:40.54 | Nilzao | [TK]D-Fender: now i know what to ask for google... CPE or NET tx |
| 17:41.42 | dshap | [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-Fender | Nilzao: 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 |
| 17:42.00 | Nilzao | dshap: i'm done here thanks |
| 17:42.14 | [TK]D-Fender | dshap: Just ask out loud unless you have special reason to ask me directly |
| 17:42.28 | [TK]D-Fender | dshap: And Just ask if you're going to ask. |
| 17:42.33 | dshap | okay well you're the one who always knows what's up around here hah |
| 17:42.34 | dshap | but yeah |
| 17:42.44 | dshap | i'm having issues passing DTMF tones through my SIP trunk it seems |
| 17:42.46 | dshap | with my Asterisk server |
| 17:43.02 | dshap | basically i'm trying to get my Asterisk server to dial an outgoing call and interact with an IVR through SendDTMFs |
| 17:43.37 | dshap | when 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-Fender | dshap: 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) |
| 17:44.01 | dshap | okay since yesterday i've narrowed down the problem quite a bit |
| 17:44.07 | Nilzao | ~ivr |
| 17:44.08 | infobot | somebody said ivr was Interactive Voice Response |
| 17:44.21 | dshap | If I do PSTN Phone --> * --> IVR |
| 17:44.21 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 17:44.31 | dshap | I can't dial DTMF tones on my phone that reach the IVR |
| 17:44.40 | dshap | they're obviously reaching my Asterisk server since I'm able to interact with my dialplan |
| 17:44.42 | Nilzao | dshap: what phone you using? |
| 17:44.51 | Nilzao | dshap: softphone? ata? |
| 17:44.52 | dshap | i've tried a cell phone and a landline |
| 17:44.55 | dshap | PSTN phone ^^ |
| 17:45.13 | Nilzao | dshap: how your PSTN phone access the *? FXS? |
| 17:45.17 | dshap | obviously when i call the IVR directly with my cell phone or landline without putting * into the equation I can interact with it no problem |
| 17:45.28 | dshap | it uses a SIP trunk |
| 17:45.38 | dshap | i call a DID which is tied to a SIP origination provider |
| 17:45.41 | dshap | my asterisk server is pure VoIP |
| 17:46.01 | Nilzao | dshap: have you tryed relaxdtmf? |
| 17:46.11 | dshap | never heard of that |
| 17:46.15 | dshap | ~relaxdtmf |
| 17:46.22 | dshap | lol neither is the info guy |
| 17:46.27 | Nilzao | dshap: its on sip.conf let me see the right name |
| 17:46.32 | BlackSlik | how do i make my first IVR with my newly install asterisk box |
| 17:46.42 | dshap | BlackSlik: lol what a question |
| 17:46.50 | dshap | BlackSlik: read the asterisk eBook, that's what i did |
| 17:46.58 | BlackSlik | which of it |
| 17:47.04 | dshap | "which of it" ? |
| 17:47.06 | ariel_ | BlackSlik: see the sample files there is a demo there |
| 17:47.32 | Nilzao | dshap: try in your sip: relaxdtmf = yes |
| 17:47.50 | Nilzao | dshap: and dtmfmode = rfc2833 |
| 17:47.51 | dshap | okay and that's under my specific trunk and not general |
| 17:47.52 | dshap | right? |
| 17:48.02 | dshap | i've tried dtmfmode=rfc2833 before but never with relaxdtmf=yes |
| 17:48.06 | dshap | i should put both in? |
| 17:48.15 | Nilzao | dshap: free to try... just a guess |
| 17:48.20 | [TK]D-Fender | dshap: No |
| 17:48.21 | Nilzao | dshap: i don't know if it will work |
| 17:48.27 | [TK]D-Fender | dshap: What phone are you using? |
| 17:48.30 | ariel_ | relaxdtmf=yes is for zap |
| 17:48.36 | [TK]D-Fender | ariel_: Not only. |
| 17:48.37 | dshap | cell phone |
| 17:48.38 | BlackSlik | how do i make my first IVR with my newly install asterisk box |
| 17:48.52 | [TK]D-Fender | dshap: ..... |
| 17:48.59 | DarthPointer | BlackSilk: if you are asking that question you may want to look into some managment software like FreePBX (not supported here, try #FreePBX) |
| 17:49.01 | dshap | cell phone --> SIP trunk --> VoIP Asterisk box --> SIP trunk --> PSTN IVR |
| 17:49.25 | [TK]D-Fender | dshap: so the lack of DTMF is jsut between your provider and *? |
| 17:49.32 | DarthPointer | BlackSilk: it can kind of pointy-clicky the whole thing up for you :) |
| 17:49.44 | dshap | [TK]D-Fender: seems like the other way around, but yes |
| 17:49.48 | [TK]D-Fender | dshap: which of those 2 providers has the issue? |
| 17:50.18 | dshap | [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 |
| 17:50.34 | dshap | maybe i should try it just for the hell of it |
| 17:50.46 | [TK]D-Fender | dshap: Do NOT test this with jsut a cell phone, that could be part of the problem. |
| 17:51.02 | dshap | okay well maybe my logic is flawed |
| 17:51.02 | [TK]D-Fender | dshap: Isolate your tests with voip.ms and use a ahrd landline phone |
| 17:51.05 | dshap | but i think that's out of the equation |
| 17:51.06 | [TK]D-Fender | sdhit is |
| 17:51.16 | dshap | if i call the IVR directly with my cell phone |
| 17:51.18 | dshap | it works |
| 17:51.24 | [TK]D-Fender | dshap: WTF IS DIRECT? |
| 17:51.31 | dshap | it has a phone number |
| 17:51.32 | dshap | i call it |
| 17:51.35 | dshap | and i can interact with it |
| 17:51.38 | [TK]D-Fender | dshap: via what? |
| 17:51.47 | dshap | touch tones |
| 17:51.52 | [TK]D-Fender | dshap: Asterisk is software and does nto pickup &#$ing MICROWAVES |
| 17:51.54 | dshap | "Press 3 for Technical Support" |
| 17:51.58 | dshap | i press 3 |
| 17:52.04 | dshap | then it says "Transferring now" |
| 17:52.19 | dshap | if i do it through my * box, it just hangs and then says "Sorry we didn't hear anything" |
| 17:52.25 | dshap | "Please make your selection" |
| 17:52.36 | [TK]D-Fender | dshap: 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 <--- |
| 17:52.45 | dshap | okay sorry |
| 17:52.46 | [TK]D-Fender | fault* |
| 17:52.48 | dshap | i'll be very upfront |
| 17:52.53 | dshap | i'm calling an AT&T voicemail backdoor number |
| 17:53.09 | dshap | if you call the number, the first thing the IVR asks you for is a 10-digit phone number |
| 17:53.13 | dshap | so you dial the phone number and press pound |
| 17:53.21 | dshap | this connects you with a user's voicemail so you can leave a message |
| 17:53.37 | dshap | my ultimate goal is to get Asterisk to leave a message on my voicemail |
| 17:53.41 | DarthPointer | BlackSilk: 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 |
| 17:53.52 | dshap | (without calling my phone) |
| 17:54.29 | dshap | when 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 |
| 17:54.37 | dshap | when i call THROUGH my * box (using Dial()) |
| 17:54.39 | xorl | jeffspeff: you recommend teliax |
| 17:54.46 | dshap | it doesn't receive my input |
| 17:54.51 | Nilzao | dshap: can you see the DTMF in your CLI? |
| 17:54.55 | dshap | yes |
| 17:54.58 | dshap | well |
| 17:54.58 | dshap | i mean |
| 17:55.02 | dshap | not as i press them on the phone |
| 17:55.04 | leifmadsen | dshap: you probably need to delay it |
| 17:55.15 | leifmadsen | logger.conf <-- enable DTMF logging |
| 17:55.16 | dshap | lefimadsen: i'm not following... |
| 17:55.22 | dshap | okay |
| 17:55.24 | leifmadsen | dshap: what is the Dial() line? |
| 17:55.36 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-5-154-128.oc.oc.cox.net) |
| 17:55.39 | dshap | Dial(SIP/8058951743@voipms) |
| 17:55.57 | leifmadsen | right... and how are you then having asterisk dial DTMF? |
| 17:56.12 | leifmadsen | or do you mean you can't even dial dtmf through asterisk manually? |
| 17:56.15 | Shaun2222 | anybody 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. |
| 17:56.18 | dshap | the latter |
| 17:56.27 | leifmadsen | dshap: your dtmf configuration is probably wrong then |
| 17:56.39 | leifmadsen | use dtmf logging to see if you're getting and sending DTMF correctly |
| 17:56.41 | Shaun2222 | it's getting annoying, just started for me, havnt changed anything |
| 17:56.46 | [TK]D-Fender | Shaun2222: there is a single momentary cut for the "beep" of CW, but that should be it |
| 17:56.52 | leifmadsen | probably set to inband when it should be out-of-band (or vice-versa) |
| 17:57.07 | dshap | leifmadsen: 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? |
| 17:57.08 | Shaun2222 | running 1.6.0.6 |
| 17:57.16 | leifmadsen | dshap: yes |
| 17:57.20 | Shaun2222 | [TK]D-Fender: is that a phone feature or *? |
| 17:57.49 | [TK]D-Fender | Shaun2222: Purely the phone |
| 17:57.50 | *** join/#asterisk stijnbe (n=stijnbe@78-21-61-204.access.telenet.be) |
| 17:57.58 | dshap | lefimadsen: 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? |
| 17:58.04 | Shaun2222 | actually 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 |
| 17:58.05 | dshap | leifmadsen: shit i spelled your name wrong |
| 17:58.25 | dshap | leifmadsen: i think the only issue is when it tries to send them back out |
| 17:58.41 | leifmadsen | dshap: no worries, everyone says it wrong too |
| 17:58.55 | leifmadsen | dshap: dtmf configuration must be wrong then |
| 17:59.02 | leifmadsen | dtmfmode=rfc2833 most likely though |
| 17:59.33 | dshap | i've got this "Cutomer Portal" web based control panel for my SIP provider |
| 17:59.36 | dshap | and they have a DTMF Mode setting |
| 17:59.39 | dshap | and right now it's on auto |
| 17:59.46 | dshap | but i could change it to RFC2833 |
| 18:00.01 | Shaun2222 | [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-Fender | Shaun2222: Yes. The same way you set the header for auto-answer for paging. Links on the WIKI |
| 18:00.47 | Shaun2222 | thanks |
| 18:01.00 | dshap | leifmadsen: where in logger.conf do i enable DTMF logging? is it just uncommenting "dtmf" under "[logfiles]" ? |
| 18:01.18 | leifmadsen | console => warning,error,notice,dtmf <-- |
| 18:03.26 | dshap | leifmadsen: 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 |
| 18:03.38 | dshap | after Dial() is called and the channels are bridged, the DTMF doesn't show up when i press on my phone |
| 18:03.58 | leifmadsen | dshap: what version of asterisk? |
| 18:04.21 | dshap | leifmadsen: 1.4.22 |
| 18:04.33 | leifmadsen | try latest 1.4 branch to see if you have the same issue |
| 18:04.46 | leifmadsen | some dtmf changes have gone in recently |
| 18:05.01 | leifmadsen | feel 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) |
| 18:05.32 | dshap | im sorry, im not sure what you're suggesting i do |
| 18:05.37 | dshap | 1.4 branch? |
| 18:05.59 | [TK]D-Fender | dshap: And please learn to test each link SEPARATELY. You keep combining all this together along with your CELL PHONE. Prove each little piece ALONE |
| 18:06.07 | carrar | leifmadsen, could you please log into his box and do it for him :) |
| 18:06.13 | leifmadsen | carrar: nope |
| 18:06.16 | leifmadsen | <-- consultant |
| 18:06.18 | carrar | heh |
| 18:06.31 | neurosys | <-- n00b |
| 18:06.34 | carrar | <-- about to take a tour through handford Nuclear site |
| 18:06.34 | dshap | [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 |
| 18:06.42 | carrar | (sp) |
| 18:06.46 | [TK]D-Fender | dshap: You have 2 providers <- |
| 18:06.52 | leifmadsen | my work here is done :) |
| 18:07.25 | dshap | [TK]D-Fender: good point, i should try to see if the problem is mine or my provider's |
| 18:07.49 | DarthPointer | <carrar> <---- Radioactive |
| 18:09.02 | ricko73 | leifmadsen: you don't expect a check in the mail for this work do you? |
| 18:09.10 | leifmadsen | ricko73: invoice has already been sent |
| 18:09.22 | leifmadsen | ricko73: s/check/cheque/ |
| 18:09.28 | ricko73 | bah |
| 18:09.32 | ricko73 | beer/bier |
| 18:09.36 | leifmadsen | totally |
| 18:09.40 | leifmadsen | s/soccer/football/ |
| 18:09.48 | carrar | I may be |
| 18:09.56 | carrar | we'll see, should be fun |
| 18:10.03 | ricko73 | hockey/hockey,eh/ |
| 18:12.21 | dshap | [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 |
| 18:12.30 | dshap | [TK]D-Fender: further, it only worked on my landline! |
| 18:14.05 | dshap | [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) |
| 18:20.49 | dshap | still 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? |
| 18:21.08 | dshap | i tried putting in a second argument for 500 ms between tones |
| 18:21.09 | dshap | still nothin |
| 18:22.57 | *** join/#asterisk propellerhead (n=yogurt2u@host205.190-230-220.telecom.net.ar) |
| 18:23.39 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-b930cba2686976ea) |
| 18:26.21 | dshap | but i'm skeptical of flowroute |
| 18:26.42 | dshap | when i call my flowroute DID, i can't even hear the Playback()s from my * box |
| 18:28.26 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:31.15 | dshap | [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? |
| 18:32.13 | carrar | dshap, did you read leif's book? |
| 18:32.18 | carrar | cover to cover |
| 18:32.43 | [TK]D-Fender | dshap: 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. |
| 18:32.54 | carrar | mmm swiss cheese |
| 18:33.08 | dshap | carrar: not cover to cover but i'm working on it |
| 18:33.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:34.03 | carrar | melted swiss cheese on toast with BEEF |
| 18:34.03 | carrar | and onions |
| 18:34.05 | DarthPointer | dshap: 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 |
| 18:35.05 | dshap | [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 |
| 18:35.13 | dshap | DarthPointer: i'll get it right up |
| 18:36.29 | dshap | here's my sip.conf http://www.pastebin.ca/1446576 |
| 18:38.49 | dshap | here's my call file and the extension that it triggers: http://www.pastebin.ca/1446578 |
| 18:41.49 | dshap | [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-Fender | dshap: Have it call your cell. Watch the call (FFS PB IT), Listen for the DTMF yourself |
| 18:44.36 | ariel_ | 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) |
| 18:45.06 | dshap | [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-Fender | dshap: You read wrong. |
| 18:45.43 | dshap | so i should be able to hear the rfc2833 signaling then? |
| 18:45.49 | [TK]D-Fender | dshap: JUST DO IT |
| 18:45.55 | dshap | doing it now |
| 18:46.13 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:49.40 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:50.06 | dshap | [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-Fender | dshap: ............. |
| 18:51.00 | dshap | [TK]D-Fender: i get the call, pick up, see "SendDTMF" show up on my CLI, and i don't hear anything |
| 18:51.14 | dshap | oh |
| 18:51.15 | dshap | PB it |
| 18:51.17 | dshap | doing that now |
| 18:52.58 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 18:52.59 | dshap | [TK]D-Fender: here it is: http://pastebin.com/d2628035f |
| 18:53.19 | dshap | [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-Fender | dshap: cOMPLETE LACK OF AUDIO IS THE PROBLEM HERE. |
| 18:54.19 | [TK]D-Fender | dshap: 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?!? |
| 18:55.04 | dshap | [TK]D-Fender: we ditched the voipms issue earlier though |
| 18:55.15 | dshap | audio works with voipms but DTMF does not |
| 18:55.26 | dshap | DTMF sort of works with flowroute but audio does not |
| 18:55.41 | [TK]D-Fender | dshap: you keep pulling these stupid surprises that waste our time. I'm tired of the runaround |
| 18:55.42 | dshap | when 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) |
| 18:56.19 | dshap | i said this earlier: <dshap> when i call my flowroute DID, i can't even hear the Playback()s from my * box |
| 18:56.33 | dshap | and the reason i said that is because i thought it was relevant |
| 18:56.59 | nullable_type | Hey 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-Fender | nullable_type: tell us what device you are testing, show us the call, and show us your configs. |
| 18:58.23 | dshap | [TK]D-Fender: flowroute *successfully* transferred DTMF from my cell phone, through my asterisk box, to the AT&T voicemail system |
| 18:58.30 | dshap | [TK]D-Fender: voipms failed to do this |
| 18:58.45 | dshap | [TK]D-Fender: voipms successfully plays audio using Playback() |
| 18:58.53 | dshap | and flowroute fails to do that |
| 18:59.01 | [TK]D-Fender | dshap: Do an ANSWER before your WAIT |
| 18:59.22 | dshap | ok |
| 19:00.31 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 19:01.28 | dshap | just tested that by having * call my cell phone with flowroute, i heard nothing at all (no DTMF or audio) |
| 19:01.57 | [TK]D-Fender | dshap: Did you see the dialplan executing only AFTER you actually picked up? |
| 19:01.58 | dshap | now trying voipms |
| 19:02.28 | dshap | [TK]D-Fender: yes that is correct. before i picked up it just said "Attempting call" on the CLI |
| 19:02.57 | dshap | i 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() |
| 19:04.03 | dshap | [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-Fender | dshap: Does the recording work? |
| 19:04.57 | Qwell | [TK]D-Fender: NAT. |
| 19:05.02 | Qwell | promise. |
| 19:05.04 | [TK]D-Fender | dshap: And do "playback(silence/2) after your Answer |
| 19:05.30 | dshap | [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-Fender | Qwell: Well we did beat this one over him about a dozen times, and the configs do look right |
| 19:08.29 | Qwell | does externip still end in .1? |
| 19:08.42 | [TK]D-Fender | Qwell: externip=70.181.88.1 |
| 19:08.49 | Qwell | winks |
| 19:09.02 | nullable_type | D-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) |
| 19:09.21 | Qwell | dshap: tell him your IP. |
| 19:09.40 | dshap | [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 |
| 19:09.41 | dshap | also |
| 19:09.45 | [TK]D-Fender | nullable_type : exten => GetDigitToConfirm,n,WaitExten() ;THIS WORKED WELL WITH EVERY OTHER PROVIDER. <- not going to F-ING trust |
| 19:09.45 | dshap | my externip is not my IP |
| 19:09.52 | dshap | it's my router's default gateway address |
| 19:10.01 | dshap | that's the only way i could get my outgoing calls to work |
| 19:10.02 | [TK]D-Fender | dshap: TWIT! |
| 19:10.07 | defsdoor | heh ? |
| 19:10.10 | [TK]D-Fender | dshap: has to be its F-ING ADDRESS |
| 19:10.31 | [TK]D-Fender | dshap: * is not asking for your NEIGHBOUR'S address, its asking for YOURS |
| 19:10.34 | defsdoor | so your other end is talking to your default gateway - not an machine you own ? |
| 19:10.35 | dshap | if i set externip=MyIP (which is DIFFERENT from my router's default gateway), then outgoing calls do not go through |
| 19:10.45 | dshap | that's when i get the circuits busy thing |
| 19:10.57 | Qwell | dshap: we went over this last night... |
| 19:11.05 | [TK]D-Fender | dshap: your router's gateway is where IT sends traffic to, but that is not ITS IP address |
| 19:11.17 | defsdoor | dshap: does your router have an internet routable IP address ? |
| 19:11.25 | Qwell | defsdoor: it does. |
| 19:11.28 | dshap | yes |
| 19:11.31 | defsdoor | use that |
| 19:11.38 | defsdoor | nothing else will work |
| 19:11.40 | [TK]D-Fender | nullable_type: -- Executing [Confirm@tpc:4] WaitExten("SIP/mytel-b6409920", "") in new stack <-- doesn't match |
| 19:11.45 | dshap | im telling u if i set it to that i'll get the circuits busy error |
| 19:11.50 | [TK]D-Fender | nullable_type: Now don't waste my time showing me partial crap |
| 19:11.55 | defsdoor | it's like sending a letter with the post office for the reply to address |
| 19:12.08 | [TK]D-Fender | defsdoor: Perfect description |
| 19:12.10 | dshap | defsdoor: i understand the concept of my router's WAN IP and Default Gateway |
| 19:12.26 | dshap | defsdoor: what i don't understand is why Asterisk only can place outgoing calls when i use the gateway number |
| 19:12.30 | [TK]D-Fender | dshap: yo need to put your router's WAN IP as "externip" |
| 19:12.41 | dshap | defsdoor: i changed it to the gateway address in an act of absolute desperation when nothing would work, and then it worked |
| 19:12.42 | defsdoor | dshap: 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 |
| 19:12.43 | ricko73 | defsdoor: 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 |
| 19:13.15 | defsdoor | dshap: and then configure your router to forward rtp to your internal address - no masqing |
| 19:13.31 | dshap | ~rtp |
| 19:13.32 | infobot | rumour 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. |
| 19:14.01 | dshap | defsdoor: if i set a DMZ on my internal address, would that effectively do what you just said? |
| 19:14.09 | defsdoor | what ? |
| 19:14.17 | Qwell | dshap: Don't do that. |
| 19:14.23 | nullable_type | D-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-Fender | Qwell: It should at least work... |
| 19:14.47 | dshap | defsdoor: 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-Fender | nullable_type: You show me 2 things that don't match and only little bits. Do you really want help? |
| 19:15.06 | nullable_type | D-Fender >> Now go fuck yourself, stop pretending like you want to help people |
| 19:15.27 | defsdoor | ooo nice |
| 19:15.36 | dshap | [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? |
| 19:15.43 | dshap | & defsdoor* |
| 19:15.44 | Qwell | dshap: Yes. |
| 19:15.53 | *** part/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 19:16.03 | ricko73 | bye bye |
| 19:16.08 | dshap | Qwell: okay understood. i am now changing externip=my actual IP from whatismyip.com |
| 19:16.08 | [TK]D-Fender | dshap: get it right or things will half-work at best, which BTW is what you're seeing here already. |
| 19:16.20 | dshap | [TK]D-fender: true |
| 19:16.27 | Qwell | what he said. |
| 19:17.27 | [TK]D-Fender | Nobody ever listens... |
| 19:17.41 | dshap | okay back to square 1 then |
| 19:17.43 | dshap | <PROTECTED> |
| 19:17.43 | dshap | [Jun 3 12:17:34] NOTICE[17570]: chan_sip.c:2953 auto_congest: Auto-congesting SIP/flowroute-0a44d180 |
| 19:17.43 | dshap | [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-Fender | Qwell: Thanks for the added insight on the IP itself... did look kinda legit, and .1 can be legal... |
| 19:17.53 | dshap | i'm assuming u want the SIP debug on the call attempt? |
| 19:18.06 | [TK]D-Fender | dshap: of course |
| 19:18.15 | defsdoor | dshap: what is your * box's default gateway ? |
| 19:18.34 | defsdoor | dshap: it /is/ the internal address of your router isn't it ? |
| 19:18.48 | dshap | yes i believe that is correct |
| 19:18.54 | defsdoor | check it |
| 19:19.02 | dshap | im not sure how to do it in linux |
| 19:19.09 | [TK]D-Fender | defsdoor: that is fine... otherwise the packets wouldn't be getting out... |
| 19:19.14 | defsdoor | netstat -r |
| 19:19.19 | [TK]D-Fender | dshap: "rounte -n" |
| 19:19.23 | [TK]D-Fender | route* |
| 19:19.30 | defsdoor | [TK]D-Fender: depends - he might have some RIP nonsense and other route out |
| 19:19.40 | InHisName | I 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) |
| 19:19.58 | InHisName | I would like to dial 3rd digit thru 12th digit |
| 19:20.01 | dshap | [root@localhost asterisk]# route -n |
| 19:20.01 | dshap | Kernel IP routing table |
| 19:20.01 | dshap | Destination Gateway Genmask Flags Metric Ref Use Iface |
| 19:20.01 | dshap | 192.168.2.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 |
| 19:20.01 | dshap | 169.254.0.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 |
| 19:20.02 | dshap | 0.0.0.0 192.168.2.1 0.0.0.0 UG 0 0 0 eth0 |
| 19:20.03 | [TK]D-Fender | InHisName: that doesn't dial anything |
| 19:20.04 | dshap | [root@localhost asterisk]# netstat -r |
| 19:20.06 | dshap | Kernel IP routing table |
| 19:20.08 | dshap | Destination Gateway Genmask Flags MSS Window irtt Iface |
| 19:20.10 | dshap | 192.168.2.0 * 255.255.255.0 U 0 0 0 eth0 |
| 19:20.12 | dshap | 169.254.0.0 * 255.255.0.0 U 0 0 0 eth0 |
| 19:20.14 | dshap | default 192.168.2.1 0.0.0.0 UG 0 0 0 eth0 |
| 19:20.17 | [TK]D-Fender | dshap: dshap STOp SPAMMING |
| 19:20.21 | dshap | sorry |
| 19:20.24 | dshap | i'll PB that |
| 19:20.25 | ricko73 | ~pastbin |
| 19:20.28 | [TK]D-Fender | InHisName: [15:20]<dshap>0.0.0.0 192.168.2.1 0.0.0.0 UG 0 0 0 eth0 <--- your gateway |
| 19:20.29 | ricko73 | ~pastebin |
| 19:20.30 | infobot | [~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 |
| 19:20.32 | defsdoor | too late now |
| 19:20.32 | [TK]D-Fender | (G) |
| 19:20.38 | dshap | k |
| 19:20.43 | dshap | right |
| 19:20.43 | kaldemar | InHisName: ${EXTEN:2:10} |
| 19:20.45 | dshap | that's what i thought |
| 19:20.45 | [TK]D-Fender | dshap: : [15:20]<dshap>0.0.0.0 192.168.2.1 0.0.0.0 UG 0 0 0 eth0 <--- your gateway |
| 19:20.48 | dshap | 192.168.2.1 |
| 19:20.50 | *** part/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 19:21.18 | InHisName | thanks kaldemar, that should work. I'll try now. |
| 19:21.47 | defsdoor | dshap: 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) |
| 19:22.26 | Qwell | defsdoor: You just described *every* "sip aware" router. |
| 19:22.32 | defsdoor | :) |
| 19:22.46 | dshap | defsdoor: Belkin F5D8230-4 v2 |
| 19:22.49 | dshap | it's a POS |
| 19:23.16 | dshap | [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) |
| 19:24.34 | InHisName | I thought this was parse string of 12 or more digits: 91XXXXXXXXXX. with . meaning "or more" Correct me if wrong..... |
| 19:25.12 | kaldemar | InHisName: . means one or more characters |
| 19:25.22 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:25.30 | [TK]D-Fender | InHisName: No, that is 12 digits + 1 or more CHARACters (13 MIN total) |
| 19:25.40 | InHisName | kaldemar: so I typed 13 or more dig by accident ? |
| 19:25.48 | hepta | i 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-Fender | hepta: Pick that actual video codecs as well |
| 19:27.00 | hepta | sorry, i didnt understand |
| 19:27.46 | hepta | i have all the 26x video codecs in the devices |
| 19:28.11 | [TK]D-Fender | hepta: in ASTERISK |
| 19:28.27 | hepta | what about it |
| 19:28.44 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 19:28.57 | [TK]D-Fender | hepta: specify your VIDEO CODECS in your PEER entries in ASTERISK or it will only offer AUDIO]\ |
| 19:29.11 | hepta | oh, as allow=? |
| 19:29.14 | [TK]D-Fender | YES |
| 19:29.27 | hepta | thanks. ill try that |
| 19:29.28 | defsdoor | allow=g729,gsm,ulaw,alaw,h263 |
| 19:29.43 | defsdoor | that's what I have for my n8[01]0 |
| 19:29.45 | hepta | so default allow is all the audio codecs of *? |
| 19:32.09 | [TK]D-Fender | hepta: There is no such thing as "default" |
| 19:32.19 | *** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 19:32.39 | hepta | what if i have al |
| 19:32.48 | hepta | allow=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) |
| 19:34.13 | dshap | anyone 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) |
| 19:34.36 | eppigy | hello |
| 19:34.38 | eppigy | i am dave |
| 19:36.39 | defsdoor | > Reliably Transmitting (no NAT) to 70.167.153.130:5060 is that correct ? |
| 19:37.52 | [TK]D-Fender | eppigy: We could see that :) |
| 19:39.19 | dshap | defsdoor: 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 |
| 19:39.39 | dshap | ~nat |
| 19:39.40 | infobot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:39.40 | dshap | oh |
| 19:39.42 | dshap | ~sipnat |
| 19:39.43 | infobot | [~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 |
| 19:40.50 | defsdoor | you got nat=yes in general though ? |
| 19:40.54 | dshap | yes |
| 19:41.15 | dshap | dshap> here's my sip.conf http://www.pastebin.ca/1446576 |
| 19:41.31 | infernix | Qwell: tried chan_mobile with android v1.5 yet? |
| 19:42.31 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:42.44 | dshap | [TK]D-Fender: anything else you'd like me to try? |
| 19:44.20 | dshap | i 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 |
| 19:47.35 | jaytee | TRABAJO! |
| 19:49.26 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:50.18 | dshap | im 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 |
| 19:50.20 | dshap | i'll be back later |
| 19:50.24 | dshap | thanks again for the help so far |
| 19:51.10 | defsdoor | # SIP/2.0 407 Proxy Authentication Required |
| 19:56.37 | defsdoor | does he need to register ? |
| 19:58.03 | [TK]D-Fender | defsdoor: No. |
| 19:58.08 | [TK]D-Fender | defsdoor: Just basi auth |
| 19:58.16 | defsdoor | it doesnt seem to be doing it |
| 19:58.39 | [TK]D-Fender | defsdoor: the "retransmit" bits indicate something else is wrong... |
| 19:59.28 | defsdoor | oh 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) |
| 20:02.51 | defsdoor | are 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) |
| 20:24.07 | dexthageek | good afternoon |
| 20:24.41 | dexthageek | I am running into a problem with Asterisk 1.6 crashing when more then one person enters a meetme conference |
| 20:25.12 | dexthageek | 1st person enters and waits with MOH |
| 20:25.24 | dexthageek | 2nd person logs in and asterisk crashes |
| 20:25.44 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 20:27.34 | dexthageek | anyone 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) |
| 20:28.48 | Katty | :> |
| 20:28.51 | Katty | :>>> |
| 20:29.26 | acxty | Hi 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/ |
| 20:29.34 | acxty | it is returning some errors |
| 20:30.03 | dexthageek | it is quiet in here, I have been waiting for a response myself |
| 20:30.26 | [TK]D-Fender | acxty: Channel: SIP/504111111 <--- nothing to call. guess you don't have a [504111111] in sip.conf |
| 20:31.55 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.16 | eppigy | hello Katty |
| 20:32.19 | eppigy | :> |
| 20:32.30 | dexthageek | I 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) |
| 20:34.18 | jaytee | Katty !!!!! *hugs* |
| 20:34.50 | juanIMP | whats the cli's output dexthageek ? have you check full messages or debug? |
| 20:35.06 | acxty | [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) |
| 20:35.17 | Katty | HAI DAVE! :> |
| 20:35.20 | Katty | hugs eppigy |
| 20:35.26 | Katty | hugs on jaytee |
| 20:35.31 | dexthageek | yup nothing coming up in full or messages |
| 20:35.36 | dexthageek | ast just core dumps |
| 20:35.53 | jaytee | acxty, [TK]D-Fender has left the building |
| 20:36.12 | jaytee | dave, where've ya been man? on vacation? |
| 20:36.24 | acxty | jaytee, may you help me with the call file? |
| 20:37.24 | jaytee | acxty, nope. not my bailiwick, cup of tea, strongsuit, etc. |
| 20:37.28 | juanIMP | while the second guy enters to the conference .... it just crashes? |
| 20:38.51 | dexthageek | yup, safe_asterisk kicks in and restarts |
| 20:39.20 | jaytee | timing source is? hardware, dahdi_dummy? |
| 20:39.39 | dexthageek | dahdi_dummy |
| 20:40.06 | jaytee | which version of 1.6.x? and what version kernel? |
| 20:40.32 | juanIMP | dmesg shows something wrong while dahdi_dummy is on ? |
| 20:41.35 | beek | acxty: Specify which context you want that to drop into. |
| 20:44.03 | dexthageek | 1.6.0.5 |
| 20:44.11 | dexthageek | we are running on EC2 |
| 20:45.20 | dexthageek | unfortunately we cannot upgrade to the latest release |
| 20:47.00 | leifmadsen | dexthageek: not much you're going to be able to figure out without a backtrace and an issue report |
| 20:47.19 | leifmadsen | see doc/backtrace.txt in your asterisk source |
| 20:47.44 | jaytee | EC2? 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) |
| 20:51.20 | dexthageek | jaytee: yes |
| 20:51.59 | jaytee | "For the love of God, Montressor!!!" |
| 20:53.12 | jaytee | dexthageek, is this a new install and a test platform? or have you had this working already with a previous version of * running on EC2? |
| 20:54.11 | dexthageek | this is our first install of Asterisk 1.6 |
| 20:54.35 | dexthageek | we have 6 ec2 instances running ast 1.4 |
| 20:55.06 | jaytee | dexthageek, running zaptel with ztdummy or dahdi with dahdi_dummy? |
| 20:55.18 | dshap | alright 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? |
| 20:57.38 | dshap | the 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) |
| 21:04.46 | dshap | [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-Fender | dshap: 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 |
| 21:06.49 | dshap | coming right up |
| 21:07.24 | eppigy | Katty: whats for din din |
| 21:07.28 | eppigy | 8[] |
| 21:07.50 | eppigy | jaytee: my shell server got rebooted |
| 21:07.58 | eppigy | it takes me time to reconnect to every network |
| 21:07.59 | eppigy | :[ |
| 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) |
| 21:12.50 | thomas | hello |
| 21:12.57 | dshap | [TK]D-Fender: here is what you requested |
| 21:13.01 | dshap | Router config: http://imagebin.ca/view/4Krlzg2.html |
| 21:13.02 | dshap | sip.conf: http://pastebin.com/m84f6426 |
| 21:13.02 | dshap | Call attempt (.call file) SIP DEBUG: http://pastebin.com/m528bcfb2 |
| 21:13.10 | *** part/#asterisk lanning (n=lanning@nat/yahoo/x-b930cba2686976ea) |
| 21:13.11 | dshap | in retrospect i probably should have made that router config private content but whatever |
| 21:13.32 | thomas | i have a little problem |
| 21:14.15 | thomas | have in my dialplan: |
| 21:14.15 | thomas | exten => _fromAmtX.,1,Dial(SIP/toAnlage${EXTEN:7}@berofix) |
| 21:14.15 | thomas | exten => _fromA[n]lageX.,1,Dial(SIP/toAmt${EXTEN:10}@berofix) |
| 21:17.30 | thomas | When i calling my number and i have a source-number then ringing my phone (all ok): http://paste.keks.be/500/txt |
| 21:17.50 | thomas | when i calling my number and i havent a source-number then: |
| 21:17.50 | thomas | [Jun 3 23:16:40] NOTICE[2630]: chan_sip.c:13879 handle_request_invite: Call from '' to extension 'fromAmt66668269' rejected because extension not found. |
| 21:18.08 | thomas | any ideas to my problem? |
| 21:18.20 | thomas | my dialplan: |
| 21:18.20 | thomas | exten => _fromAmtX.,1,Dial(SIP/toAnlage${EXTEN:7}@berofix) |
| 21:18.21 | thomas | exten => _fromA[n]lageX.,1,Dial(SIP/toAmt${EXTEN:10}@berofix) |
| 21:19.07 | dshap | [TK]D-Fender: this also: http://imagebin.ca/view/dJB5nN.html |
| 21:20.45 | [TK]D-Fender | dshap: ..... |
| 21:20.53 | [TK]D-Fender | dshap: WTF is RTP?! |
| 21:20.53 | dshap | what else do you need? |
| 21:21.02 | thomas | [TK]D-Fender: hello. excuse me, but, can you help me with my problem? |
| 21:21.10 | dshap | ~rtp |
| 21:21.11 | infobot | [rtp] The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 21:21.22 | dshap | okay so i need to forward another port is what you're saying |
| 21:21.24 | [TK]D-Fender | dshap: WHERE the &#^$ is it? |
| 21:21.34 | [TK]D-Fender | dshap: You didn't forward the ports for RTP <_ |
| 21:21.35 | dshap | i dunno |
| 21:21.41 | [TK]D-Fender | ......... |
| 21:21.41 | eppigy | oh boy |
| 21:21.48 | dshap | okay im sorry RTP is a new concept that i just learned today |
| 21:21.48 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 21:21.49 | dshap | hah |
| 21:21.54 | dshap | what ports? |
| 21:22.01 | [TK]D-Fender | ~sipnat |
| 21:22.02 | infobot | [~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 | ^^^^^^^^^^^^ |
| 21:22.38 | dshap | http://www.voip-info.org/wiki/view/RTP+Ports |
| 21:22.44 | dshap | RTP: UDP ports 16384-32767 |
| 21:22.46 | dshap | i should probably forward those |
| 21:23.03 | [TK]D-Fender | dshap: I've linked you the guide about a dozen times. READ THAT ONE |
| 21:23.09 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 21:24.24 | dshap | i've read that guide |
| 21:24.27 | dshap | i think i'm #3 |
| 21:24.28 | Katty | eppigy: idunno what's for dindin :< |
| 21:24.37 | Katty | eppigy: that requires work. i'm all worked out |
| 21:24.49 | dshap | the 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-Fender | dshap: 1st f-ing link |
| 21:25.26 | dshap | shit i think i overread the RTP thing on that first link |
| 21:25.28 | dshap | ughh |
| 21:25.55 | thomas | can anyone help me ? |
| 21:26.26 | jaytee | eppigy, I missed ya man! you've been gone for days seems like |
| 21:26.29 | *** join/#asterisk Shinu (n=Shinu@unaffiliated/shinu) |
| 21:27.08 | eppigy | :D |
| 21:27.14 | eppigy | yes it has been a minute |
| 21:27.25 | eppigy | Katty: well I will make din din then |
| 21:27.40 | eppigy | plots a course to Boston Market |
| 21:27.44 | Katty | eppigy: k |
| 21:28.58 | dshap | [TK]D-Fender: for some reason my router needs to restart to apply port forward changes |
| 21:28.59 | dshap | brb |
| 21:29.08 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 21:29.14 | jaytee | I 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. |
| 21:30.16 | jaytee | anyway, it's quittin time. bbiab |
| 21:30.48 | AlmightyOatmeal | [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? |
| 21:31.36 | AlmightyOatmeal | when 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.. |
| 21:31.47 | AlmightyOatmeal | i'm not really sure what i didn't do right :( |
| 21:32.04 | jaytee | ever 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? :-) |
| 21:32.39 | AlmightyOatmeal | was that geared toward me? |
| 21:33.02 | jaytee | yes, MrOmnipotentHighFiberCereal |
| 21:33.45 | AlmightyOatmeal | jaytee: thats an incoming call from a landline though.. i thought it would bring up a menu prompt or ring directly to extension 100 :( |
| 21:34.06 | Katty | yawns sleepily |
| 21:34.17 | Katty | falls asleep somewhere, randomly |
| 21:34.30 | jaytee | hmm, 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. |
| 21:34.43 | AlmightyOatmeal | i guess i need to work on my dialplans :) |
| 21:34.47 | AlmightyOatmeal | jaytee: 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) |
| 21:35.31 | dshap | [TK]D-Fender: now I have the entire RTP port range forwarded to my * box |
| 21:35.47 | jaytee | and excuse the offhand responses, my buttocks doth contain great wisdom that verily I oft find impossible to contain! :-) |
| 21:35.50 | dshap | did not fix my problem |
| 21:36.16 | jaytee | in simple english, I'm a wiseass! |
| 21:36.28 | jaytee | back later all |
| 21:36.30 | jaytee | peace |
| 21:37.00 | AlmightyOatmeal | lol |
| 21:38.11 | thomas | If 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: |
| 21:38.25 | thomas | NOTICE[2630]: chan_sip.c:13879 handle_request_invite: Call from '' to extension 'fromAmt66668269' rejected because extension not found. |
| 21:38.28 | thomas | any ideas? |
| 21:39.39 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:39.41 | acxty | beek, I want to make a outgoing call, specifically a cellphone |
| 21:40.51 | lancey | thomas: you sure you have such extension called 'fromAmt66668269' o_O? |
| 21:41.46 | acxty | [TK]D-Fender, I fix what you told me earlier. I am getting this error know http://dpaste.com/51161/ |
| 21:41.53 | thomas | lancey: when i have set a sourcenumber. http://paste.keks.be/501/txt |
| 21:42.20 | *** join/#asterisk korihor (n=korihor@190.72.237.209) |
| 21:42.56 | lancey | this seems to be calling different extension? |
| 21:43.15 | [TK]D-Fender | acxty: and I don't see SIP debug in there, and no configs |
| 21:43.39 | [TK]D-Fender | dshap: "type=peer", remove the "insecure" line. |
| 21:43.55 | acxty | [TK]D-Fender, only sip.conf or something else? |
| 21:44.08 | [TK]D-Fender | acxty: sip.conf |
| 21:44.16 | afink | guys I just moved an asterisk box and I get this when trying to start zaptel: http://pastebin.com/m3b672fda |
| 21:44.27 | acxty | ok |
| 21:44.40 | AlmightyOatmeal | when 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? |
| 21:44.47 | *** join/#asterisk BlackSlik (n=james@41.219.213.219) |
| 21:45.21 | [TK]D-Fender | AlmightyOatmeal: "the number for the box"? Huh? |
| 21:45.51 | AlmightyOatmeal | [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. |
| 21:46.05 | acxty | [TK]D-Fender, here it is http://dpaste.com/51187/ |
| 21:46.33 | *** join/#asterisk dshap (n=dshap@ip70-181-91-110.oc.oc.cox.net) |
| 21:46.42 | dshap | that was weird - my router just dropped wifi |
| 21:46.45 | dshap | im back now though |
| 21:46.47 | dshap | *sigh* |
| 21:47.38 | AlmightyOatmeal | [TK]D-Fender: what did i screw up? heh :( |
| 21:48.21 | [TK]D-Fender | AlmightyOatmeal: 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-Fender | dshap: JUST fyi, YOUR ROUTER, BEING THE pos THAT IT IS, COULD BE THE ENTIRE CULPRIT HERE. |
| 21:49.26 | AlmightyOatmeal | [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 |
| 21:49.27 | dshap | [TK]D-Fender: later im going to connect my server directly to my modem and see if i can send the DTMF |
| 21:50.13 | acxty | [TK]D-Fender, do you found something wrong on the sip.conf file? |
| 21:51.11 | [TK]D-Fender | AlmightyOatmeal: 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-Fender | acxty: Your call file is not calling a target NUMBER there |
| 21:52.30 | AlmightyOatmeal | [TK]D-Fender: so i need to create an extension with the phone number? hum ok |
| 21:53.04 | acxty | [TK]D-Fender, I have Extension: on it |
| 21:53.10 | [TK]D-Fender | actShow me |
| 21:53.21 | acxty | do I need to add that extension on the sip.conf? |
| 21:53.42 | [TK]D-Fender | actNO. the CHANNEL does not tell your PROVIDER what # to call. Or is that some fixed enpoint like a SIP phone? |
| 21:54.13 | acxty | [TK]D-Fender, no, pure asterisk |
| 21:54.22 | [TK]D-Fender | acxty: Who are you calling? |
| 21:54.36 | acxty | I am trying to make a call from asterisk to a cellphone |
| 21:55.00 | [TK]D-Fender | acxty: What cellphone? |
| 21:55.09 | [TK]D-Fender | acxty: I don't see another cell # in there |
| 21:55.09 | acxty | my |
| 21:55.40 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:56.12 | AlmightyOatmeal | i can't set an extension as long as my phone number ugh.. i dont get it |
| 21:56.19 | acxty | [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-Fender | acxty: 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-Fender | it* |
| 21:58.45 | [TK]D-Fender | AlmightyOatmeal: Pardon? Why the hell not? |
| 21:58.49 | acxty | [TK]D-Fender, SIP/50411111/99999999 |
| 21:59.16 | [TK]D-Fender | acxty: Yes, that looks more normal |
| 21:59.23 | AlmightyOatmeal | [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 |
| 21:59.30 | AlmightyOatmeal | its a little frustrating for me until i learn more |
| 22:00.32 | scooby2 | Is there a way to log the number someone called in the CDR? |
| 22:00.48 | scooby2 | I strangely remember this working in 1.2 but I just noticed in 1.4 it is not. |
| 22:00.52 | dshap | [TK]D-Fender: the RTP port forwarding made SendDTMF() work with my voip.ms provider!!!! |
| 22:00.58 | acxty | [TK]D-Fender, if I change that, how does the Extension: will go? |
| 22:04.00 | acxty | [TK]D-Fender, I set extension 0, it is working know thanks ;) |
| 22:04.57 | AlmightyOatmeal | so i can't make a call either: |
| 22:05.02 | AlmightyOatmeal | [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. |
| 22:05.06 | AlmightyOatmeal | i just don't understand |
| 22:05.17 | scooby2 | I see the problem. For some reason src and dst in cdr are being input into mysql as both being the src # |
| 22:06.09 | dshap | [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 |
| 22:06.26 | dshap | [TK]D-Fender: although the externip issue remains a mystery |
| 22:06.28 | [TK]D-Fender | dshap: I was afraid of that ;) |
| 22:06.29 | lancey | AlmightyOatmeal: in your sip.conf/iax.conf, wherever you defined 'jamie' you have also specified a context |
| 22:06.34 | dshap | haha |
| 22:06.38 | lancey | you don't have a '60blabla' extension in it. |
| 22:06.54 | lancey | or anything else that would match this (.e.g. _X., or 6X. etc) |
| 22:07.39 | AlmightyOatmeal | uh, i should |
| 22:07.42 | AlmightyOatmeal | 2 sec |
| 22:08.09 | AlmightyOatmeal | lancey: if you don't mind, http://pastebin.ca/1446829 are the extensions i have |
| 22:08.40 | lancey | and what's the context of 'jamie' ? |
| 22:08.47 | [TK]D-Fender | AlmightyOatmeal: Now aside from all the other stuff I'd blast you for... what in there should macth 6083994252? |
| 22:08.57 | lancey | well, nothing of these matches the number you dialed - 6xxxx |
| 22:09.14 | lancey | you also have a missing bracket (>) on the last line |
| 22:09.22 | AlmightyOatmeal | lancey: the context to jamie is my default extensions that i just pasted |
| 22:09.25 | [TK]D-Fender | lancey: and the context you referred to... well.. I don't even want to think where you came up with that idea :) |
| 22:09.37 | AlmightyOatmeal | [TK]D-Fender: i'm going directly from what BroadVoice said :( |
| 22:09.40 | lancey | [TK]D-Fender? |
| 22:10.19 | [TK]D-Fender | AlmightyOatmeal: Forget what they say, and look at what they DO. |
| 22:10.27 | lancey | scratches his head |
| 22:10.35 | lancey | [TK]D-Fender would you mind being more clear about that, please? |
| 22:10.42 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:10.44 | AlmightyOatmeal | so i should change _1NxxNxxxxxx to my phone number? |
| 22:11.00 | [TK]D-Fender | AlmightyOatmeal: 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. |
| 22:11.11 | lancey | AlmightyOatmeal: generally, you should have at least TWO contexts |
| 22:11.32 | [TK]D-Fender | AlmightyOatmeal: You are showing us an INBOUND call attempt from BV, correct? |
| 22:11.52 | AlmightyOatmeal | [TK]D-Fender: that last one was outbound, i pasted inbound earlier |
| 22:11.58 | lancey | [TK]D-Fender: lol. did he call his provider 'jamie' |
| 22:11.59 | lancey | :) |
| 22:12.20 | [TK]D-Fender | lancey: No, thats just the CALLERID of the caller |
| 22:13.25 | [TK]D-Fender | AlmightyOatmeal: try some new PB's |
| 22:13.34 | AlmightyOatmeal | PB's? |
| 22:13.37 | [TK]D-Fender | ~pb |
| 22:13.38 | infobot | [~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/ |
| 22:13.49 | AlmightyOatmeal | of? |
| 22:14.25 | [TK]D-Fender | AlmightyOatmeal: 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 |
| 22:17.25 | *** join/#asterisk SuD (n=alex@89.140.32.2.static.user.ono.com) |
| 22:17.28 | *** part/#asterisk SuD (n=alex@89.140.32.2.static.user.ono.com) |
| 22:17.33 | *** join/#asterisk SuD (n=alex@89.140.32.2.static.user.ono.com) |
| 22:18.41 | AlmightyOatmeal | [TK]D-Fender: http://pastebin.ca/1446844 <-- that? |
| 22:20.21 | [TK]D-Fender | AlmightyOatmeal: go to * CLI and ENABLE SIP DEBUG "sip set debug on" and try again. |
| 22:25.28 | AlmightyOatmeal | [TK]D-Fender: http://pastebin.ca/1446853 <-- when i dial in |
| 22:26.21 | [TK]D-Fender | AlmightyOatmeal: TOO LATE. You show it only AFTER the failure |
| 22:26.31 | AlmightyOatmeal | uh |
| 22:26.34 | [TK]D-Fender | AlmightyOatmeal: ENTIRE damn call please can configs along-with |
| 22:26.49 | AlmightyOatmeal | .. |
| 22:27.05 | [TK]D-Fender | s/can/and/ |
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| 22:27.52 | *** part/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:30.39 | AlmightyOatmeal | [TK]D-Fender: http://pastebin.com/f14e254df and configs: http://pastebin.ca/1446844 |
| 22:30.54 | AlmightyOatmeal | unless you want the whole config file |
| 22:33.15 | [TK]D-Fender | AlmightyOatmeal: 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 |
| 22:33.17 | infobot | [~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 |
| 22:33.20 | VaGoNeTaS | [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-Fender | AlmightyOatmeal: ^^^^^ follow the guide |
| 22:34.13 | VaGoNeTaS | in our office we dont have analogue lines anymore, only VoIP |
| 22:34.13 | AlmightyOatmeal | [TK]D-Fender: ok |
| 22:34.16 | VaGoNeTaS | i heard something about HylaFax, , is that possible? or i still need an ATA modem or something? |
| 22:34.57 | pfn | I forget, what's the process for exposing a blocked callerid? forward it to an 800 number? |
| 22:35.10 | [TK]D-Fender | AlmightyOatmeal: 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-Fender | AlmightyOatmeal: 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-Fender | AlmightyOatmeal: that is the point of contexts, to separate things. |
| 22:39.05 | AlmightyOatmeal | so noted |
| 22:39.27 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:40.23 | AlmightyOatmeal | [TK]D-Fender: then i need to learn more about extensions for handling inbound calls |
| 22:40.30 | AlmightyOatmeal | and same for outbound heh |
| 22:40.36 | pfn | ugh, why does teliax charge for inbound calls, not cheap |
| 22:42.24 | AlmightyOatmeal | well outbound calls seem to work |
| 22:42.34 | AlmightyOatmeal | i'll read more on extensions tonight |
| 22:42.54 | AlmightyOatmeal | thanks for the guidance [TK]D-Fender |
| 22:43.46 | [TK]D-Fender | AlmightyOatmeal: Dialplan = 95% of * |
| 22:44.10 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 22:44.54 | jaytee | anyone ever tried using PC-6400 800mhz DDR2 dimms on a mobo that will only accept up to PC-5400 667mhz DDR2? |
| 22:47.10 | AlmightyOatmeal | jaytee: normally the faster ram would simply downclock, not normally a risk to stability |
| 22:49.22 | jaytee | that'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 |
| 22:50.03 | scooby2 | [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. |
| 22:50.07 | jaytee | I'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 |
| 22:50.26 | jaytee | think I'll give it a shot. hopefully be back in just a few :-) |
| 23:05.27 | KavanS | anyone have a suggestion for a good voip client for mac? |
| 23:05.41 | pfn | so, if teliax charges for inbound minutes does it mean they should be required to expose ani/clid? |
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| 23:22.01 | seb- | [TK]D-Fender: sorry my Ekiga has been so cranky....got any time @home now? |
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| 23:52.14 | Typhoeus_ | 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. |
| 23:52.25 | Typhoeus_ | Sorry should of used a ? :) |
| 23:52.27 | *** join/#asterisk johnakabean (n=john@static-173-50-101-10.nrflva.east.verizon.net) |
| 23:52.55 | johnakabean | hey everyone; i'm having a problem with asterisk. using freepbx 2.5.1 and asterisk.1.4.22, I type amportal start |
| 23:52.59 | johnakabean | asterisk starts FINE |
| 23:53.03 | johnakabean | and runs for about an hour |
| 23:53.14 | johnakabean | then I get an error "automatically restarting asterisk" in console |
| 23:53.40 | Typhoeus_ | did you look at the "full" log to see if it shows why it is restarting? |
| 23:53.46 | johnakabean | one sce |
| 23:55.04 | dshap | How 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()? |
| 23:56.15 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:57.21 | Typhoeus_ | dshap: Are you trying to do like a voice broadcast to a list of numbers? |
| 23:58.20 | generalhan | so, 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 ? |
| 23:58.21 | dshap | Typhoeus_: 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 |
| 23:58.49 | dshap | unless that is the appropriate way to do it.. |