| 00:01.58 | snadge | done.. i sent simon a short polite email with a few questions in it |
| 00:02.06 | vinhdizzo | [TK]D-Fender: http://pastebin.com/XdQyrcBi |
| 00:02.27 | snadge | one of the questions was a bit open ended.. would he consider using LVS sip persistence engine over netscaler in a production environment :p |
| 00:02.59 | vinhdizzo | [TK]D-Fender: not sure why i'm getting silence when the call is being transferred. in asterisk 1.8, i get a ringing tone. |
| 00:03.48 | snadge | tk: how many mbits of traffic do you manage at peak? (assuming you're a sysadmin who looks after asterisk servers) :p |
| 00:04.24 | snadge | or is that a personal question.. kinda like.. how big is your doodle |
| 00:05.49 | Kobaz | i like to make big doodles |
| 00:05.58 | [TK]D-Fender | vinhdizzo: Looks like you are coming in GV, and then right back out...\ |
| 00:06.19 | [TK]D-Fender | vinhdizzo: This is not a "transfer". This is a completely seperate second call. |
| 00:06.25 | [TK]D-Fender | vinhdizzo: * is merely bridging |
| 00:06.39 | vinhdizzo | i see |
| 00:06.47 | vinhdizzo | is this not the recommended method? |
| 00:06.55 | vinhdizzo | is there a way to completely pass on the call? |
| 00:07.07 | vinhdizzo | without asterisk bridging the call? |
| 00:07.20 | snadge | i see.. no comment :P |
| 00:07.58 | [TK]D-Fender | vinhdizzo: And from the indication issue, I'd say taht * has no way to indicate "ringing", etc across that channel-type |
| 00:08.09 | [TK]D-Fender | vinhdizzo: So if you expect to ehar something you'd have to answer first |
| 00:08.50 | vinhdizzo | i see. so in 1.8, things were handled a different way that it rang? |
| 00:10.50 | [TK]D-Fender | I'm not going to guess on it.... |
| 00:11.07 | vinhdizzo | ok |
| 00:11.29 | vinhdizzo | they way im 'transferring' call, is that the best way to do it? |
| 00:13.00 | vinhdizzo | I'm using a Dial statement |
| 00:14.25 | [TK]D-Fender | That isn't a transfer. |
| 00:14.28 | [TK]D-Fender | Never call it one. |
| 00:14.32 | [TK]D-Fender | * ssits in the middle |
| 00:14.37 | [TK]D-Fender | ~b2bua |
| 00:14.37 | infobot | hmm... b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 00:14.58 | [TK]D-Fender | there is no "hand-off" for Motif either. |
| 00:15.10 | [TK]D-Fender | Does GV even ALLOW you 2 channels at a time? |
| 00:15.21 | vinhdizzo | not sure |
| 00:15.47 | [TK]D-Fender | vinhdizzo: I've seen some comments that suggest this might not be the case |
| 00:15.52 | vinhdizzo | ok |
| 00:16.10 | [TK]D-Fender | vinhdizzo: I'd recommend doing tests aside from your calling right back out. |
| 00:16.33 | vinhdizzo | how do i do that? |
| 00:16.55 | vinhdizzo | at any rate, since im using a Dial statement |
| 00:17.04 | vinhdizzo | is there a way to play music until the other end picks up? |
| 00:17.23 | Penguin | core show application Dial |
| 00:17.27 | vinhdizzo | would appreciate it if you could point me to a link |
| 00:17.59 | vinhdizzo | or some search terms |
| 00:18.17 | [TK]D-Fender | vinhdizzo: dial something ELSE |
| 00:18.26 | [TK]D-Fender | vinhdizzo: Stop going in Motif and right back out |
| 00:18.31 | [TK]D-Fender | use a SIP phone and go out motif |
| 00:18.38 | [TK]D-Fender | come in via motif and dial a SIP phone |
| 00:19.17 | vinhdizzo | ahh. i wasn't able to set up sip in asterisk yet |
| 00:40.12 | snadge | bazinga.. god bless the SIP rfc ;) |
| 00:40.25 | snadge | i hope its accurate |
| 00:40.59 | snadge | In a new request created by a UAC outside of any dialog, the Call-ID |
| 00:41.00 | snadge | <PROTECTED> |
| 00:41.00 | snadge | <PROTECTED> |
| 00:41.00 | snadge | <PROTECTED> |
| 00:41.06 | snadge | oops that wasnt meant to multiline |
| 00:43.30 | snadge | It SHOULD be the same in each registration from a UA. |
| 00:45.05 | snadge | i think that's a bit vague.. do i need to look at the asterisk source code to see if the Call-ID field is unique per call connection attempt from a single client? |
| 00:51.56 | vinhdizzo | thanks [TK]D-Fender, navaismo, and redotis for your help today! |
| 00:52.51 | snadge | my basic understanding of that, is that call-id should be the same for each UA / registration.. but the cseq is incremented to differentiate dialogs |
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| 00:54.24 | file | nope. |
| 00:54.39 | file | the call-id remains the same for the registration, each individual call has its own call-id |
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| 00:55.55 | snadge | excellent.. this lends really well to L7 type load balancing then |
| 00:56.13 | snadge | its like they already thought of that when they designed the protocol :P |
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| 00:58.07 | snadge | in an ideal world, you wouldn't have to look inside packets to make routing decisions |
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| 03:01.25 | drmessano | OMG its file |
| 03:01.35 | drmessano | Wait, 3 hours ago |
| 03:01.39 | drmessano | Darn timestamps |
| 03:03.11 | drmessano | ERROR[5934]: chan_motif.c:2035 jingle_interpret_google_transport: Received Google transport information on session 'XXXXXXXXXXXX@10.76.75.7' but ICE support not available <--- Any thoughts? |
| 03:03.31 | drmessano | ICE is enabled in rtp.conf (though it should be on by default, eh?) |
| 03:08.22 | pabelanger | drmessano, module show like pjsip |
| 03:10.07 | drmessano | pabelanger, 0 |
| 03:10.14 | pabelanger | that is your issue |
| 03:10.16 | drmessano | 11.6 rc1 btw |
| 03:10.19 | pabelanger | you need pjproject |
| 03:10.24 | drmessano | Ok.. |
| 03:10.39 | pabelanger | ~pjproject |
| 03:11.36 | drmessano | I thought that was bundled with the 11 source and compiled with Asterisk... First time i've run into this in 11.x |
| 03:12.26 | pabelanger | no |
| 03:12.29 | pabelanger | it was removed |
| 03:12.48 | pabelanger | so, now you need to fetch it from http://github.com/asterisk/pjproject |
| 03:12.56 | drmessano | Ahhhh.. |
| 03:17.12 | drmessano | Installing now.. thank you pabelanger |
| 03:18.14 | pabelanger | np |
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| 03:41.21 | drmessano | pabelanger, any other ideas. I followed the pjsip directions on the wiki. ./configure in Asterisk picked up nothing different, so no change on make |
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| 03:57.28 | pabelanger | you followed https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject |
| 03:59.44 | drmessano | yep |
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| 04:06.32 | [TK]D-Fender | hates his power supply.... |
| 04:14.58 | jeev | shoot it |
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| 05:13.12 | ChannelZ | why is more of pjsip being written into 11? |
| 05:15.52 | coppice | I guess they've turned pjsip up to 11 |
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| 05:54.02 | whtsup | hello |
| 05:54.19 | whtsup | can anyone help me in jitter buffer for iax2 |
| 05:54.56 | whtsup | my delay between 2 server is 150 ms but sometimes it went to 300ms |
| 05:55.09 | whtsup | wht will be the jitter buffer setting for iax2 |
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| 06:16.51 | _omer | hello |
| 06:17.11 | _omer | I have a simple question: How does PHPAGI communicates with ASterisk ? through AMI ? |
| 06:22.03 | snadge | is anyone successfully using an L7 balancing solution in front of asterisk? |
| 06:23.50 | snadge | i need to talk to a consultant about such things.. is this something that digium could perhaps help with? |
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| 14:30.22 | vandyk | I've been using FreePBX distro for years, now I think that I'm facing some issues with it. I always used FreePBX GUI to configure Asterisk, but I need a clean Asterisk environment to isolate some problems. I know that there is AsteriskNow. Someone knows if AsteriskNow generates clean dialplans? |
| 14:32.08 | vandyk | or if there is a GUI that is "clean" regarding .conf files? |
| 14:35.27 | vandyk | when I say AsteriskNow, I'm referring about its GUI |
| 14:40.14 | [TK]D-Fender | vandyk: No. |
| 14:40.25 | [TK]D-Fender | vandyk: GUI's don't write friendly interactive code |
| 14:40.46 | [TK]D-Fender | vandyk: So what are the actual issues you're facing? |
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| 17:17.37 | file | hum hum hum |
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| 18:18.32 | vandyk | D-Fender: remember that issue about calls being canceled? I'm suspecting that is something about FreePBX. everytime I receive a call on Asterisk, it will come with a suffix. This suffix tells me that I need to search the dialed number against a mysql table. If I don't find any records, I need to check in a webservice, then with the webservice return, I know the correct number to dial |
| 18:19.17 | [TK]D-Fender | vandyk: no |
| 18:19.36 | [TK]D-Fender | vandyk: Your comms had started. There is no "databse lookup" with FreePBX there. There is no dialplan happening there. |
| 18:19.45 | tm1000 | vandyk: AsteriskNOW uses freepbx |
| 18:19.52 | tm1000 | there is no different gui there anymore |
| 18:19.55 | tm1000 | hasnt been for years |
| 18:20.25 | tm1000 | [TK]D-Fender: +1 (as always) |
| 18:20.35 | vandyk | D-Fender: I know that there is no database lookup on FreePBX. I developed PHP code to do that. |
| 18:21.04 | tm1000 | vandyk: if you want plain asterisk then thats what you have to do. start with nothing |
| 18:21.07 | [TK]D-Fender | tm1000: vandyk Still not where this is going... you are in the MIDDLE of a a dial-out. There is NO place for you to start executing code there |
| 18:21.24 | tm1000 | [TK]D-Fender: im agreeing with you |
| 18:21.37 | [TK]D-Fender | tm1000: Sorry, accidentally had your auto-complete tehre in front |
| 18:21.39 | [TK]D-Fender | accident |
| 18:21.49 | tm1000 | :-) |
| 18:23.50 | vandyk | in fact the actual setup seems to work, most of times, at least on my tests dialing from an extension, it does the job. |
| 18:24.23 | vandyk | But when I have an overload of calls (call center dialler), I think that things are not going well |
| 18:24.34 | vandyk | I was looking this morning a SIP trace |
| 18:24.50 | vandyk | and I found two distinct calls to the same number |
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| 18:25.13 | [TK]D-Fender | vandyk: Your description is getting vaguer and vaguer and we don't have anything to comment on here.... |
| 18:25.26 | vandyk | one call with the suffix added and another without the suffix |
| 18:26.47 | [TK]D-Fender | I don't recally any of these "suffixes", and am not sure what their purpose is expected to be here.... |
| 18:29.36 | vandyk | I'll try explain the whole scenario. I receive calls from a dialler caming from a call center. All calls are to mobile numbers. I need to identify the mobile operator and then send to the correct GSM gateway to complete the call. I'm using Asterisk with FreePBX. On extensions_custom.conf I have this: |
| 18:29.39 | vandyk | [from-internal-custom] |
| 18:29.39 | vandyk | exten => _000099.,1,AGI(checktrunk.php); |
| 18:29.40 | vandyk | exten => _000099.,n,Hangup(); |
| 18:31.09 | vandyk | the checktrunk.php is an AGI that will lookup in mysql database if there is mobile operator there. If it is found, it will remove the 000099 from the dialed number and add the mobile operator code in front of number to be dialed |
| 18:31.24 | vandyk | then on FreePBX I have outbound routes set |
| 18:31.33 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 18:31.46 | vandyk | with mobile operator code + number to send to GSM gateways |
| 18:32.22 | vandyk | each gsm gateway has configuration to each mobile operator |
| 18:32.52 | [TK]D-Fender | vandyk: the only side I recall you showing was an outbound call |
| 18:33.01 | [TK]D-Fender | vandyk: nothing inbound at all |
| 18:33.19 | vandyk | so, the logic should be this: none calls starting with the 000099 would be send to the GSM gateways |
| 18:33.38 | [TK]D-Fender | vandyk: And "call center" could mean a lot of things. |
| 18:34.11 | vandyk | when I mention call center, I'm referring a dialler that send me about 50 calls at same time |
| 18:35.40 | [TK]D-Fender | What dialer? |
| 18:35.54 | [TK]D-Fender | tell you what .. jsut get new debug... |
| 18:35.57 | [TK]D-Fender | we need something real to look at |
| 18:36.09 | vandyk | I have a debug with sip |
| 18:36.15 | vandyk | I'll put on pastebin now |
| 18:40.51 | vandyk | Pastebin will not allow to put the whole debug |
| 18:40.56 | vandyk | I uploaded the file |
| 18:41.00 | vandyk | http://www.sendspace.com/file/2lxeu6 |
| 18:41.17 | vandyk | please do a search for the number: 80268994 |
| 18:41.35 | vandyk | then you'll see the start of call |
| 18:42.19 | vandyk | INVITE sip:00009980268994@ |
| 18:49.43 | [TK]D-Fender | vandyk: So I see you passing "183 Sessions progress back" for the call and it is at that instant that the calling end decides to kill the call. |
| 18:49.53 | [TK]D-Fender | vandyk: You are way outside of AGI or anything else as I said. |
| 18:50.00 | [TK]D-Fender | You are in the midst of an actual dial. |
| 18:51.21 | vandyk | yes, I'm in the middle of an actual dial, correct. |
| 18:51.54 | vandyk | this means that it can't be done what I'm doing? |
| 18:52.22 | [TK]D-Fender | no. |
| 18:52.32 | [TK]D-Fender | Said it before, saying it again |
| 18:53.01 | [TK]D-Fender | You: "Here's progress!". Gateway :"GTFO!" |
| 18:55.48 | vandyk | but why this works in some cases and others don't? When I implemented this I was thinking about the same principle of ENUM trunks |
| 18:56.10 | [TK]D-Fender | I'd have to see that "other case" |
| 18:56.21 | vandyk | ENUM trunks will check if there is a sip endpoint number and redirect the call to it |
| 18:56.21 | [TK]D-Fender | You need to give specific examples we can compare |
| 18:56.39 | vandyk | you mean one call that was completed with success? |
| 18:56.55 | [TK]D-Fender | enum doesn't "just happen". You are using a trunk definition |
| 18:57.37 | vandyk | please find the number 91322039 |
| 18:57.49 | vandyk | this is a call that was completed successfully |
| 19:02.44 | [TK]D-Fender | not sure of the difference |
| 19:03.16 | [TK]D-Fender | but it is SIP/CentralPortocred that is cancelling after you send back progress |
| 19:04.35 | vandyk | That's why I'm confused. I've been running this system over 5 months without problems |
| 19:04.43 | vandyk | 2 weeks ago |
| 19:04.49 | vandyk | started if this issues |
| 19:05.05 | vandyk | the only thing changed: the amount of simultaneous calls |
| 19:05.47 | *** join/#asterisk boratynskikamil (5311a3f3@gateway/web/freenode/ip.83.17.163.243) |
| 19:06.15 | boratynskikamil | Hello. How should I write pattern for extension to make sure that called phone end at 43? _N.43? |
| 19:06.29 | boratynskikamil | ends* |
| 19:09.13 | [TK]D-Fender | boratynskikamil: there is no such pattern to check for "ends with" |
| 19:09.26 | [TK]D-Fender | boratynskikamil: You'll have to accept a wider pattern and validate it in the dialplan itself |
| 19:12.20 | Penguin | If you enter digits after N or X, why won't it still match when the number matches the pattern? |
| 19:14.44 | Penguin | Such as _NXXXXX43 |
| 19:15.16 | [TK]D-Fender | only works if the pattern length is fixed |
| 19:15.25 | boratynskikamil | As Fender said. |
| 19:15.25 | Penguin | Agreed. |
| 19:15.34 | Penguin | You need variable length? |
| 19:15.38 | [TK]D-Fender | boratynskikamil: At which point I've answered the other end of it |
| 19:16.11 | boratynskikamil | Ok, so one more question. |
| 19:16.22 | boratynskikamil | How could I set up calling number? |
| 19:17.06 | [TK]D-Fender | boratynskikamil: your question is unclear |
| 19:19.04 | boratynskikamil | I may call from 2 SIPs at the moment. How could I specify that SPI1 will call from *46 end and SP2 from 43? |
| 19:19.41 | *** join/#asterisk nikzadi123 (~fateme@192.211.48.88) |
| 19:20.22 | [TK]D-Fender | boratynskikamil: You term "from" is unclear |
| 19:20.44 | [TK]D-Fender | and a "SIP" is not a thing |
| 19:20.58 | boratynskikamil | It is protocol, I know. |
| 19:21.08 | [TK]D-Fender | be clear about what SERIVE or DEVICE is caling IN to your server... and what you are trying to dial OUT |
| 19:22.01 | boratynskikamil | I mean, a peer, would be better term, right? |
| 19:23.19 | [TK]D-Fender | Still vague |
| 19:23.23 | [TK]D-Fender | Everything could be a peer |
| 19:23.45 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:23.46 | [TK]D-Fender | the fact you use one for a SIP phone and the other might be fore a SIP ITSP is not implicit |
| 19:23.58 | [TK]D-Fender | be VERY specific about what is coming IN vs OUT |
| 19:26.21 | boratynskikamil | You mean: what am I calling from and where? |
| 19:27.38 | [TK]D-Fender | boratynskikamil: Give a new better description |
| 19:27.42 | [TK]D-Fender | boprstart over |
| 19:27.48 | [TK]D-Fender | boratynskikamil: start over |
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| 19:32.09 | boratynskikamil | So, I have 2 PCs. |
| 19:32.13 | boratynskikamil | Softphones, right? |
| 19:34.09 | boratynskikamil | May I specify that one of them is calling from diffrent number (I have 3 numbers in the bank). |
| 19:36.21 | Penguin | You must have an idea that callerid numbers are significant. |
| 19:41.29 | boratynskikamil | Penguin - yep. |
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| 20:03.34 | boratynskikamil | Ok, thank you guys. As always, you helped me. |
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