| 00:01.25 | *** join/#asterisk ChadAragorn (~ChadArago@206.251.40.221) |
| 00:07.41 | no2pencil | thank you for your time WIMPy |
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| 01:35.12 | janicez | Interesting problem. I've got "1 1/2 way voice". The downstream via EDGE works perfectly despite CG-NAT, but the upstream makes me sound "like a Dalek." (stuttering) |
| 01:35.36 | janicez | Is it the CPE that's the problem, or the PBX? |
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| 01:38.04 | janicez | ChadAragorn: Might you know what is going on here? I've got "1 1/2 way voice". The downstream via EDGE works perfectly despite CG-NAT, but the upstream makes me sound "like a Dalek." (stuttering) |
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| 02:08.02 | WIMPy | janicez: Too little bandwidth? |
| 02:08.34 | janicez | WIMPy: Probably. :/ |
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| 02:10.36 | WIMPy | Try a codec with more reduction. |
| 02:11.04 | janicez | I'm literally using the lowest bandwidth codec my SIP client supports. And it's still rip. |
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| 02:12.28 | WIMPy | Well, VOIP needs a perfect network. |
| 02:12.56 | janicez | Correct. :( |
| 02:13.33 | WIMPy | You could try VOIP via dialup *eg* |
| 02:17.30 | janicez | WIMPy: hahaha |
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| 02:38.13 | phix | WIMPY!!!!!!!!!!!!1111 |
| 02:38.43 | WIMPy | ยก |
| 02:42.29 | phix | "D |
| 02:42.34 | phix | s/"/:/ |
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| 04:20.13 | mattsl | So I have static agents in a queue that are logging in and out by just logging out of their softphones. This, correctly, puts them in a state of Unavailable in the queue. However, if they use DND at any point, then they forever more show Not in use in the queue even if their phone is no longer registered. |
| 04:20.22 | mattsl | Any ideas on why this is happening and how to fix it? |
| 04:25.09 | [TK]D-Fender | mattsl, that DND is CUSTOM stuff... this isn't an * problem yet |
| 04:25.21 | [TK]D-Fender | and by custom .... that clearly means FreePBX |
| 04:25.24 | janicez | good god |
| 04:25.27 | [TK]D-Fender | Keep it in there :) |
| 04:26.54 | mattsl | [TK]D-Fender: Was trying blame lorsungcu. ;-) |
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| 07:27.21 | esupra | good day |
| 07:27.26 | spicyramen_ | good day |
| 07:27.54 | esupra | I have an issue that hopefully somebody can help with, or had in the past |
| 07:28.03 | esupra | asterisk 11 on ubuntu, apt-get installed |
| 07:28.06 | esupra | segfault at fffffffffffffffe ip 00007f3548d81e7e sp 00007f352c1c5cb0 error 5 in res_config_odbc.so[7f3548d7f000+8000] |
| 07:28.15 | esupra | with odbc get that error in syslog and asterisk dies |
| 07:28.50 | esupra | mysql db |
| 07:29.45 | esupra | on res_mysql i get the following : ERROR[11460] res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. |
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| 07:46.39 | kds46 | Hi all |
| 07:46.44 | esupra | hi |
| 07:47.01 | kds46 | it is first day to incomming IRC of Asterisk |
| 07:48.23 | kds46 | Because I have a big problem for the setting Asterisk, I come here!! |
| 07:48.40 | esupra | what is the problem |
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| 07:49.53 | kds46 | wait a moment, Let me have time to prepare for explain this problem. :) |
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| 07:59.12 | kds46 | I want to consist as below system. |
| 07:59.29 | kds46 | phone ---> opensips ---> asterisk 13 |
| 07:59.50 | kds46 | so I use the Dispatcher Module in Opensips. |
| 08:00.26 | kds46 | Opensips's Dispatcher send OPTIONS Request to Asterisk(for sip ping) |
| 08:00.55 | kds46 | But Asterisk responses 401. |
| 08:01.40 | kds46 | so I make the SIP Trunk on Asterisk, and then Asterisk respones 200 OK for OPTIONS Request. |
| 08:02.12 | kds46 | I am not sure tha this way is correct or not |
| 08:02.53 | kds46 | and then Phone send REGISTER Request to Asterisk via Opensips. |
| 08:04.01 | kds46 | But Asterisk response 404. when Phone send REGISTER Request to Asterisk directly, it's fine( asterisk response 200) |
| 08:04.19 | kds46 | I have no idea about that now |
| 08:05.02 | kds46 | CLI Log is "WARNING[8171]: res_pjsip_registrar.c:680 registrar_on_rx_request: AOR '1002' not found for endpoint 'opensips_in_1'" |
| 08:05.28 | kds46 | 'opensips_in_1' is the SIP Trunk Name on Asterisk. |
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| 08:36.40 | mirela666 | kds46, try using canreinvite=no or nat=yes, seems to me that asterisk box can't reach phone directly and might be tring |
| 08:37.07 | mirela666 | kds46, for opensips_in_1 trunk |
| 08:37.23 | kds46 | ok, I will do it now... |
| 08:41.57 | kds46 | hmmm, these are already set as "canreinvite=no and nat=yes". |
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| 09:25.37 | mirela666 | kds46, maybe you can see for opensips side of configuration |
| 09:26.14 | esupra | nat=port,comedia |
| 09:26.43 | kds46 | OK, thank you... |
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| 11:54.31 | Demon_VoIP | Hi. Problem: asterisk 13.4. user account with the same settings and credentials are successfully registered in the case chan_sip and 403 Forbidden in case of res_pjsip. |
| 11:54.32 | Demon_VoIP | The difference in SIP packages only no "algorithm=MD5" in string Authorization in the case of res_pjsip. Is there a way to forcibly add it there? |
| 11:57.22 | file | what is the device? |
| 11:58.05 | Demon_VoIP | asterisk 13.4 invound registration with "Server: Smile CTI Server" (intertelecom.ua) |
| 11:58.24 | file | so, you're talking outbound registration |
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| 11:59.40 | Demon_VoIP | Apparently, I confused all the time. As a result "pjsip show registation" : Rejected |
| 12:00.07 | file | without modifying the code there's no way to get it in there if it isn't there, and I'm not sure if that would be us or PJSIP to modify |
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| 12:00.33 | Demon_VoIP | pjprotect 2.4. i think... sip_auth_client.c |
| 12:00.56 | file | entirely possible |
| 12:01.29 | Demon_VoIP | Well. I got the answer. Whatever it was. Thanks |
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| 14:39.34 | i0x81 | Hey, does anyone know why the +1 would get prepended to the caller id that is set in the asterisk, Set(CALLERID(num)= is used to set the number without +1 at the front, however the remote party still sees it being prepended |
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| 14:42.52 | WIMPy | If you don't set it, it must be someone else. |
| 14:43.09 | WIMPy | It won't happen by itself. |
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| 14:50.49 | i0x81 | WIMPy: im guessing its done by the sip provider |
| 14:52.02 | WIMPy | That would make sense. |
| 14:52.40 | WIMPy | Unless it's internal, you sohouldn't ever use any other format. |
| 14:55.25 | i0x81 | Anybody knows anything about caller id backspoofing ? |
| 14:55.40 | WIMPy | back? |
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| 15:01.27 | i0x81 | WIMPy: this stuff http://www.binrev.com/forums/index.php/topic/41175-backspoofing-with-asterisk/ |
| 15:02.02 | i0x81 | You basically attempt to pull out the CNAM records like name for a specific called id |
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| 15:09.37 | WIMPy | Do you want to download the database that way? |
| 15:14.32 | i0x81 | not sure |
| 15:14.38 | i0x81 | was hoping someone knew how it worked |
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| 17:18.14 | i0x81 | anybody has dynamic caller id setup by using dtmf |
| 17:18.15 | i0x81 | ? |
| 17:23.25 | [TK]D-Fender | What do you be "setup by using dtmf"? |
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| 17:31.52 | cmendes0101 | Like hit a prompt to enter a callerid then dial out? |
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| 17:34.37 | i0x81 | where youd call in, type in digits into your headset, system would parse them as fake caller id and destination, set the caller id and patch you through to the destination |
| 17:35.25 | i0x81 | cmendes0101: exactly |
| 17:36.46 | cmendes0101 | i0x81: Well you just do a Read() to get DTMF from the user |
| 17:37.29 | i0x81 | i guess then youd just do setCallerID and dial the parsed var |
| 17:38.42 | cmendes0101 | If your on 1.2 or 1.4 since setCallerid is depricated. Use Set(CALLERID(num)=xxx) |
| 17:45.08 | i0x81 | looks like the version im using is 11.18 |
| 17:45.22 | i0x81 | cmendes0101: do you know if there is a limit on the Set(CALLERID(all) |
| 17:45.56 | cmendes0101 | i0x81: like? |
| 17:46.49 | i0x81 | eh, sorry i meant character limit |
| 17:46.57 | i0x81 | not sure if its my client or not |
| 17:47.06 | cmendes0101 | If you are using a voip provider, they have the final say on the callerid |
| 17:47.10 | i0x81 | when i pass 26 char string, it no longer shows up on the caller id |
| 17:47.18 | cmendes0101 | I'm not sure on char limit but maybe check the wiki |
| 17:47.27 | i0x81 | gonna have to run a sniffer and see if its reaching the client |
| 17:47.32 | cmendes0101 | wiki.asterisk.org |
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| 18:54.40 | i0x81 | Does anybody know where this username parameter is set in asterisk |
| 18:54.40 | i0x81 | From: "BLAH BLAH +1123456789" <sip:asterisk@111.111.111.111:5060>;tag=fgdfgfdgfd |
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| 18:54.54 | i0x81 | the "asterisk@111.111.111.111" asterisk part |
| 19:01.43 | newtonr | i0x81, chan_sip or chan_pjsip? |
| 19:01.50 | i0x81 | er, looks like its set with Set(CALLERID(num), if you dont explicity set it, it defaults to "asterisk" |
| 19:03.00 | newtonr | for requests to an endpoint, you can set from_user and from_domain for the endpoint |
| 19:03.03 | newtonr | (chan_pjsip) |
| 19:04.00 | newtonr | CALLERID(num) is going to set the number part of "BLAH BLAH +1123456789" in your example |
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| 19:43.22 | Micc_ | Why do I still get AMI Newexten events when I don't have dialplan set for my user? |
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| 19:51.41 | sweettea | http://packages.asterisk.org/centos/6/asterisk-11-certified/ |
| 19:51.44 | sweettea | anyone know why this is empty? |
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| 19:54.14 | Frojoe | Does anyone know anything about 2b channel transfer? |
| 19:54.32 | Frojoe | Because I have a weird issue regarding it |
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| 19:59.57 | TazzNZ | Frojoe, what is the issue ? |
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| 20:10.48 | sweettea | where are good asterisk repos at? |
| 20:10.54 | sweettea | i really dont want to compile |
| 20:13.30 | i0x81 | for which repo ? |
| 20:13.36 | i0x81 | distro* |
| 20:14.23 | Frojoe | TazzNZ, It seems that one of our PRI lines, whenever we attempt to start a 2b channel transfer, will randomly start sending a 2 byte invoke ID to the telco switch. Our telco only accepts 1 byte invoke IDs |
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| 20:18.36 | TazzNZ | Frojoe, I guess you have taken this up with the telco ? |
| 20:18.44 | Frojoe | Yup |
| 20:18.52 | Frojoe | S'how we found out what the issue was |
| 20:18.55 | TazzNZ | and the are blaming Asterisk |
| 20:19.21 | TazzNZ | oh right - I misread that - *you* are sending the 2 byte ID |
| 20:20.36 | TazzNZ | what cards are you using ? |
| 20:27.23 | Frojoe | TazzNZ, let me find ou t |
| 20:27.27 | Frojoe | This got handed off to me :P |
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| 20:47.59 | Frojoe | TazzNZ, Digium TE410P REV F QUAD E1/T1 CARD w/ VPMOCT128 Echo Cancellation Module REV B |
| 20:48.03 | Frojoe | Sorry for the delay |
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| 20:58.46 | sweettea | i0x81: anything certified. prefer 11 > 1.8 > 13 |
| 20:59.52 | TazzNZ | Frojoe, I would suggest that you log a bug/issue with digium, given they create the card,driver and asterisk :) |
| 21:00.09 | TazzNZ | and they are hard core - I have had nothing but joy working with them |
| 21:00.27 | Frojoe | TazzNZ, that was the next step. Figured I'd check and see if anyone here knows anything |
| 21:00.30 | Frojoe | Thanks though |
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| 21:20.26 | [TK]D-Fender | sweettea, Bad order there |
| 21:20.31 | [TK]D-Fender | and 1.8 should not even be on the list |
| 21:20.54 | mjordan | is a fan of 13, but he is biased |
| 21:23.09 | sweettea | [TK]D-Fender: okay. well Im on an old 1.8 setup. I'd like to setup a new box using only centos repo binaries. Suggestions? |
| 21:26.12 | [TK]D-Fender | 1.8 = DOA |
| 21:26.19 | [TK]D-Fender | 13 is as stable as 11. |
| 21:26.35 | [TK]D-Fender | And it's supported longer going forward and adds a lot of functionality |
| 21:26.48 | [TK]D-Fender | no real reason not to simply move to 13 |
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| 21:51.02 | WIMPy | Frojoe: Somds more like you need to fix your telco. |
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| 22:42.08 | phix | sweettea: why centos? |
| 22:42.29 | phix | Mornin' WIMPy and [TK]D-Fender! |
| 22:42.52 | WIMPy | Good morning phix |
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| 23:17.59 | sweettea | phix: what else? |
| 23:18.33 | sweettea | [TK]D-Fender: okay, but where can I find the cert repo of any? |
| 23:20.14 | [TK]D-Fender | https://www.digium.com/products/asterisk/certified-asterisk |
| 23:20.20 | [TK]D-Fender | Cert offer you virtually northing |
| 23:20.30 | [TK]D-Fender | It is only more direct support for paying members |
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| 23:20.49 | [TK]D-Fender | And isn't updated as often which may include other benificial things you'll lose out on. |
| 23:20.56 | [TK]D-Fender | SLOW updates = suck quite often |
| 23:21.18 | [TK]D-Fender | What is the benefit of using Certified Asterisk if I'm not a Digium customer? |
| 23:21.19 | [TK]D-Fender | There are no special benefits to using Certified Asterisk for users or customers that are not under an SLA with Digium. If your needs as an open source user align with those of SLA customers, you may use Certified Asterisk under the terms of the GPLv2, just as with standard open source releases of Asterisk. |
| 23:21.28 | [TK]D-Fender | Helps when you read the big print on this |
| 23:21.45 | [TK]D-Fender | So do youself a favour and forget "Cert" |
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| 23:26.21 | sweettea | hmmm |
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| 23:39.31 | sweettea | ill make the jump to 13 then |
| 23:43.23 | phix | WIMPy: How's your day been? |
| 23:43.43 | phix | sweettea: Well anything based off Debian (including Debian) comes to mind. |
| 23:44.09 | dtrainor_ | Just to learn a little bit more, can someone link me to an explanation on what the parts of CDR(clid) actually are? e.g. "dtrainor" <1111>, can I change the formatting of that, what if I only need the '1111' part etc. |
| 23:44.38 | WIMPy | Good. 2nd tuesday of the month is our data travellers / developers regulars table. |
| 23:45.02 | [TK]D-Fender | dtrainor_, only 2 parts. Name & number |
| 23:45.04 | WIMPy | thinks that Debian is the bastard operating system from hell. |
| 23:45.45 | dtrainor_ | ah ok |
| 23:46.47 | dtrainor_ | Would the "number" part be the same as the first occurance of ${EXTEN}? |
| 23:47.35 | WIMPy | The called number usually has no name, but the calling number does. |
| 23:49.14 | dtrainor_ | understood, makes sense |
| 23:49.49 | dtrainor_ | i suppose the right thing to do is just to programatically extract the "number" part out of the caller ID string |
| 23:52.43 | newtonr | dtrainor_, https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information some of that will help |
| 23:58.45 | Micc_ | Is there still no way to set the outbound leg caller id other than from the extension dialed? |
| 23:59.16 | WIMPy | There are many ways. |