| 00:01.42 | Georger | ok |
| 00:02.00 | Georger | i have centos |
| 00:02.24 | Georger | current config of snmpd was working in past for 1.4 asterisk |
| 00:03.10 | Georger | i installed from repositories asterisk-snmp and enabled the subagent |
| 00:03.40 | Georger | snmp is working in machine but cannot fetch info about asterisk |
| 00:04.32 | *** join/#asterisk jgowdy (~jgowdy@basharteg.com) |
| 00:05.18 | Georger | snmpwalk -v 2c -c public 127.0.0.1 ASTERISK-MIB::astVersionString |
| 00:05.20 | Georger | ASTERISK-MIB::astVersionString = No Such Object available on this agent at this OID |
| 00:06.27 | Georger | and.. |
| 00:06.32 | Georger | server*CLI> module show like snmp |
| 00:06.32 | Georger | Module Description Use Count |
| 00:06.32 | Georger | res_snmp.so SNMP [Sub]Agent for Asterisk 0 |
| 00:06.32 | Georger | 1 modules loaded |
| 00:07.15 | Georger | asterisk-mib.txt and digium-mib.txt also copied |
| 00:07.56 | Georger | and snmpd.conf has the lines: |
| 00:07.59 | Georger | master agentx |
| 00:08.00 | Georger | agentXSocket /var/agentx/master |
| 00:08.00 | Georger | agentXPerms 0660 0550 nobody asterisk |
| 00:29.39 | Georger | ..anybody? |
| 01:01.29 | Neptu | hej how can i see the sip accounts i created using sip reload?? |
| 01:23.13 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 01:25.42 | gladier | Neptu: you dont create sip accounts by reloading - all it does it re-read the config files. "sip show peers" will show you all configured peers |
| 01:33.30 | *** join/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 01:33.35 | *** part/#asterisk wesphillips (~wphill04@c-76-30-173-12.hsd1.tx.comcast.net) |
| 01:48.59 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 02:02.39 | Neptu | why im always getting more things in my dialplan than expected.... http://pastebin.com/wc3FXVjc i have not defined nothing more than extensions.conf and deleted a lot of other conf files but i still have some garbage there... |
| 02:09.04 | p3nguin | neptu: Pastebin your entire extensions.conf. |
| 02:12.10 | Neptu | p3nguin: quite minimalistic only have a [default] and thats it... just testing |
| 02:12.43 | p3nguin | Great, pastebin the entire thing. |
| 02:29.05 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:35.42 | *** join/#asterisk dandate2 (~dan@222.127.53.129) |
| 02:35.56 | dandate2 | does anyone know why my cisco 7940 ip phones use 2x more bandwidth than my cisco 7905 ip phones? |
| 02:36.30 | dandate2 | all i can think is mabye the 7940 has a super high bitrate |
| 02:36.39 | carrar | Cause you are using a different codec |
| 02:36.46 | dandate2 | they are both ulaw |
| 02:36.57 | carrar | what speeds are you seeing? |
| 02:37.07 | carrar | should be about 88kbps for a single call |
| 02:37.26 | dandate2 | the 7940 is using 100kb with tunneling and vpn features enabled, the 7905 only about 64kb with the same tunnelling and vpn |
| 02:37.44 | carrar | 7940 doesn't have tunneling and vpn features |
| 02:37.57 | dandate2 | right thats setup within the catalyst switch |
| 02:38.10 | dandate2 | ip tunnels and vpn to the * box |
| 02:38.14 | carrar | Sounds like you got something else going on |
| 02:38.33 | carrar | and probably not using the codec you think |
| 02:39.13 | carrar | obviously your total bw at 64kbps with tunneling, and overhead is not g711 |
| 02:39.44 | dandate2 | what could it be? sip show channels shows ulaw when the 7905s are connected |
| 02:40.58 | dandate2 | we have a half T1 line with 756kb max load. 8 connected 7940's are running 800kb load, and 10 7905s are well under the 756kb max |
| 02:46.20 | p3nguin | I've never used a 7905, but I do use 7940s and 7960s, and they don't use more bandwidth than should be expected for a call. |
| 02:46.37 | carrar | I suspect he's running g.729 |
| 02:46.41 | carrar | as the numbers add up for that |
| 02:47.06 | p3nguin | It should be easy enough to look to find out. |
| 02:47.13 | carrar | You'd think |
| 02:48.59 | dandate2 | do the cisco phones have controllable bitrate? |
| 02:50.47 | *** join/#asterisk albertoandrade (~albertoan@201.21.137.31) |
| 02:50.48 | carrar | they have selectable codecs |
| 02:51.20 | p3nguin | ulaw is 64 kbits/s. |
| 02:51.36 | carrar | (without overhead) |
| 02:52.05 | p3nguin | With overhead, I was seeing almost 90 kbits/s usage. |
| 02:52.19 | p3nguin | That's a single call leg, one call. |
| 02:53.58 | p3nguin | I just measured one call leg again. 79.3Kb |
| 02:54.35 | p3nguin | That's according to iftop. |
| 02:55.02 | p3nguin | It reads higher if I measure with iptraf. |
| 02:59.13 | carrar | dandate2: https://www.osburn.com/ss/Screen%20Shot%202012-02-25%20at%206.56.25%20PM.jpg |
| 02:59.19 | carrar | see the far right column |
| 03:00.16 | dandate2 | yeah |
| 03:00.31 | dandate2 | it honestly seems like its using g729, but i only have one license for that and cli reports ulaw usage |
| 03:00.59 | p3nguin | You won't be using ulaw and using any less bandwidth than what it uses for everyone. |
| 03:01.17 | dandate2 | ill go turn it all back on and force ulaw on each extention to see what happens |
| 03:07.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:12.14 | *** join/#asterisk RickCogley (~anonymous@EM1-113-92-162.pool.e-mobile.ne.jp) |
| 03:18.45 | dandate2 | so the 7905 reports a 50kb peak bandwidth in iftop |
| 03:18.56 | dandate2 | but ulaw is forced on the extention |
| 03:24.41 | dandate2 | i figured it out |
| 03:24.46 | dandate2 | the 7905 is using silence suppression |
| 03:29.31 | p3nguin | If you turn that off, it jumps up to where the other phones are? |
| 03:30.42 | dandate2 | yeah |
| 03:30.47 | p3nguin | I thought silence suppression provided comfort noise, suppressing any silence that may exist. |
| 03:31.14 | dandate2 | it seems to make it emit bandwidth in measures of a few hundred bytes when noones talking |
| 03:39.14 | *** join/#asterisk neurosys_ (~neurosys@c-98-254-216-32.hsd1.fl.comcast.net) |
| 03:47.57 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 04:03.06 | *** join/#asterisk jetlag (~jetlag@pool-71-168-250-69.cmdnnj.east.verizon.net) |
| 04:08.10 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001599.dsl.bell.ca) |
| 04:08.26 | volga629 | What does mean can't call outbound We are requesting SRTP, but they responded without it! |
| 04:30.15 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 04:38.03 | volga629 | How to disable srtp globally ? |
| 04:48.49 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 05:24.44 | *** join/#asterisk Subrosian (~Bot2@c-98-211-226-95.hsd1.fl.comcast.net) |
| 05:28.04 | Subrosian | i'm working on a project that that allows the customer to pull up their ticket information by entering their ticket number and the system reads back the update to them. I don't even know what this is called to be able to start planning. Similar to how you can find out how much money you have in a bank account using bank-by-phone. Would anyone know where to point me? |
| 05:28.22 | *** join/#asterisk RickCogley (~anonymous@EM1-113-92-162.pool.e-mobile.ne.jp) |
| 05:44.39 | [TK]D-Fender | Subrosian: this is basic IRV and Read() input |
| 05:44.41 | [TK]D-Fender | ~book |
| 05:44.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:44.43 | [TK]D-Fender | ^^^^ |
| 05:44.54 | [TK]D-Fender | Time to read up on how you can control your calls |
| 05:46.00 | Subrosian | this talks about how to pull information from external sources? |
| 05:46.41 | volga629 | I am trying make work snom 370 with srtp can't make certificates happy Problem setting up ssl connection: error:14094416:SSL routines:SSL3_READ_BYTES:sslv3 alert certificate unknown |
| 05:48.03 | [TK]D-Fender | Subrosian: no that would be using DB calls, etc |
| 05:48.13 | [TK]D-Fender | Subrosian: depending where this information is stored |
| 05:49.05 | Subrosian | do you have any reference material for doing this in particular? Pulling from web-page and parsing? or maybe Mysql? |
| 05:52.23 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 06:04.01 | *** join/#asterisk twomashi (~Adium@186.19.170.164) |
| 06:04.28 | twomashi | Hi all. I'm having a problem setting a variable to the output of a script, though I think the issue is to do wtih the output itself. |
| 06:04.44 | twomashi | Is it possible that output containing special characters would not be read properly? |
| 06:04.59 | twomashi | exten => 111,1,Set(SONG=${SHELL(/etc/asterisk/scripts/randmp3.sh):0:-1}) |
| 06:08.37 | twomashi | Or.. appears that it's timing out waiting for the script to output something, even though it takes <1s |
| 06:09.06 | twomashi | ah no… Its a permissions issue :) |
| 06:09.11 | twomashi | nevermind! |
| 06:09.31 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 06:29.09 | volga629 | SRTP unprotect: authentication failure where I can find proper patch for this issue I so issue reported in version 1.8.2 |
| 06:29.24 | volga629 | and my in use 1.8.7 |
| 06:56.13 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 07:16.06 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 07:16.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 07:17.05 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 07:29.20 | volga629 | How to right way apply patch for asterisk patch.gz ? |
| 07:29.35 | volga629 | need gzip first and apply patch ? |
| 07:35.45 | *** join/#asterisk nitram (foo@korben.fhloston.org) |
| 08:10.51 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:24.12 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 08:25.55 | p3nguin | volga629: gzip -cd patch.gz | patch -p0 |
| 08:36.43 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 08:37.59 | WIMPy | thinks there is a < missing. |
| 08:55.32 | volga629 | thank that work |
| 08:55.42 | volga629 | here another error Unable to open vm-messagex2 (format 0x100 (g729)): No such file or directory |
| 08:56.30 | volga629 | this for russian language pack is available some info how to fix it ? |
| 08:57.46 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 08:59.24 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 09:02.07 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001599.dsl.bell.ca) |
| 09:10.11 | p3nguin | wimpy: No, there wasn't. |
| 09:55.37 | *** join/#asterisk RickCogley_ (~anonymous@114.179.8.185) |
| 10:11.26 | Georger | hi i posted also yesterday about snmp but no answer,anyone familiar here ? |
| 10:21.34 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 11:28.58 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:16.08 | *** join/#asterisk jakent (~john@c-24-125-38-65.hsd1.va.comcast.net) |
| 12:31.52 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 12:48.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 13:07.42 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:35.20 | *** join/#asterisk ketas-ts (~ketas@62.65.215.151.cable.starman.ee) |
| 13:45.03 | *** join/#asterisk Georger (~Georger@79.103.153.7.dsl.dyn.forthnet.gr) |
| 13:56.06 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:14.03 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 14:34.57 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 14:35.19 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 14:35.20 | *** join/#asterisk freeedrich|_ (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 14:52.03 | tzanger | dang, where's coppice when you need him. :-) |
| 15:06.04 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:21.02 | *** join/#asterisk SirLouen (sirlouen@84.125.178.248.dyn.user.ono.com) |
| 15:21.17 | SirLouen | hi all |
| 15:21.39 | *** join/#asterisk briankb (~briankb@pool-98-119-173-84.lsanca.fios.verizon.net) |
| 15:41.46 | Georger | anyone familiar with snmp? |
| 15:44.12 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 15:49.17 | SirLouen | if I have a SIP trunk with a DID associated, how can I change the default context for inbound calls, for other context? the only solution I've found is including the context property in the [general] from sip.conf |
| 15:51.06 | [TK]D-Fender | SirLouen: "sip trunk" is a poor term. They should be matching a peer which specifies is own context for the calls to land in. As for the call itself, your DID arrives at the provider. how they pass it to you is another matter. Most dial it in in the INVITE which you'd just make an exten to match |
| 15:51.25 | [TK]D-Fender | SirLouen: And if it's bypassin your peer and falling to [general] then fix your peer |
| 15:53.35 | SirLouen | [TK]D-Fender truth, but I match a peer with its own context then match an exten with no success |
| 15:53.50 | SirLouen | in the CLI i see it goes to the default dialplan context |
| 15:54.13 | [TK]D-Fender | then it isn't matching your peer |
| 15:54.13 | SirLouen | I figured out it was taking the "default" context from the [general] tab in the sip.conf |
| 15:54.52 | [TK]D-Fender | enable SIP debug when you're looking at these calls |
| 15:54.59 | SirLouen | and thats why I got that solution, by creating a context property into the [general] tab |
| 15:55.08 | [TK]D-Fender | not good... |
| 15:55.17 | [TK]D-Fender | look at your peer and the call.... |
| 15:56.00 | SirLouen | i'm using asterisk realtime for the peer |
| 15:56.18 | SirLouen | and the register line under the [general] in the sip.conf |
| 15:57.04 | bbourdage | If I would like voicemail to be sent via email to 2 people, how would I do that ?, I know that I can put Vm&VM, and have it save to two boxes, but I only want it to save to 1 mailbox, but email 2 people. I have tried putting in 2 email address's, seperated by commas, and ";", but the system fails. Is setting an Alias the only way ? |
| 15:57.07 | Dovid | anyone here use cepstral in an AGI? |
| 15:58.36 | [TK]D-Fender | bbourdage: You can't you need to do a distribution list on the MTA on your server, or have that on the client side |
| 15:59.07 | [TK]D-Fender | bbourdage: OR... replace the sendmail binary that * calls to do the split. |
| 15:59.18 | SirLouen | is actually the context property or maybe the regcontext property? |
| 15:59.28 | [TK]D-Fender | context <- |
| 15:59.47 | bbourdage | Thanks TK, that is what I thought, was hoping to keep the total config in MySQL. |
| 16:00.20 | [TK]D-Fender | bbourdage: Well... I think you could put it in the field and replace the script so that it parses it out for you... |
| 16:00.40 | [TK]D-Fender | bbourdage: Which you would then modify the binary * calls to clean it up |
| 16:01.25 | bbourdage | TK - Thanks, That is a great idea |
| 16:01.41 | [TK]D-Fender | bbourdage: It's all about finding the right spot in the process to cheat :) |
| 16:02.22 | SirLouen | [TK]D-Fender found the solution here: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 16:02.27 | [TK]D-Fender | plays pool using "suggestions" of physics ... |
| 16:02.30 | SirLouen | "Another Example" |
| 16:02.39 | [TK]D-Fender | SirLouen: doesn't really tell me much.... |
| 16:03.05 | SirLouen | if you see, it puts the context property in the [general] tab to solve the problem |
| 16:03.26 | [TK]D-Fender | .... |
| 16:03.40 | [TK]D-Fender | SirLouen: No. That doesn't not solve the problem. That patches the flaw |
| 16:03.46 | [TK]D-Fender | Doesn't solve* |
| 16:04.12 | [TK]D-Fender | I shouldn't not never ever not stop using double negatives Like I don't anymore...... |
| 16:04.41 | [TK]D-Fender | watches CleverBot implode from the back-calculating.... |
| 16:05.48 | SirLouen | lol |
| 16:05.53 | SirLouen | yeah, i believe s |
| 16:05.56 | SirLouen | believe so |
| 16:06.07 | SirLouen | you think the sip provider is not sending sip headers properly? |
| 16:06.31 | [TK]D-Fender | SirLouen: I think I don't see your configs or SIP debug and trust neither inherently :) |
| 16:29.54 | *** join/#asterisk bmg505 (~leon@196-209-123-82.dynamic.isadsl.co.za) |
| 16:38.26 | *** join/#asterisk Georger (~Georger@79.103.153.7.dsl.dyn.forthnet.gr) |
| 16:39.53 | Georger | pasting my issue |
| 16:39.57 | Georger | i have centos |
| 16:39.57 | Georger | current config of snmpd was working in past for 1.4 asterisk |
| 16:39.57 | Georger | i installed from repositories asterisk-snmp and enabled the subagent |
| 16:39.57 | Georger | snmp is working in machine but cannot fetch info about asterisk |
| 16:40.08 | Georger | snmpwalk -v 2c -c public 127.0.0.1 ASTERISK-MIB::astVersionString |
| 16:40.08 | Georger | ASTERISK-MIB::astVersionString = No Such Object available on this agent at this OID |
| 16:40.18 | Georger | server*CLI> module show like snmp |
| 16:40.19 | Georger | Module Description Use Count |
| 16:40.19 | Georger | res_snmp.so SNMP [Sub]Agent for Asterisk 0 |
| 16:40.19 | Georger | 1 modules loaded |
| 16:40.26 | Georger | asterisk-mib.txt and digium-mib.txt also copied |
| 16:40.26 | Georger | and snmpd.conf has the lines: |
| 16:40.26 | Georger | master agentx |
| 16:40.26 | Georger | agentXSocket /var/agentx/master |
| 16:40.26 | Georger | agentXPerms 0660 0550 nobody asterisk |
| 16:40.44 | Georger | if anyone can help i will appriciate it |
| 16:44.44 | WIMPy | Georger: I hve never managed to get a reply, either. |
| 16:45.14 | volga629 | Unable to open vm-messagex2 (format 0x100 (g729)): No such file or directory |
| 16:45.31 | Georger | WIMPy for me seems like the agentx is not working |
| 16:45.47 | WIMPy | I've got it standalone. |
| 16:46.15 | volga629 | this russian language sounds pack |
| 16:46.47 | Georger | needs to be subagent so i can monitor full machine |
| 16:47.16 | Georger | should i go to asterisk-dev? |
| 16:47.54 | WIMPy | It's not really the interntion. |
| 16:48.30 | WIMPy | And weekends are generelly a bad time to ask. |
| 16:49.04 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:51.14 | coppice | tzafrir: are you around? |
| 17:08.26 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:11.08 | volga629 | Asterisk ended with exit status 139 what is mean ? |
| 17:18.11 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 17:19.14 | ChannelZ | Probably a not-too-good crash |
| 17:19.25 | volga629 | yes |
| 17:20.28 | volga629 | http://fpaste.org/1bRa/ |
| 17:20.55 | volga629 | I am trying set SRTP, but not really working |
| 17:21.19 | volga629 | this is snom phone 370 |
| 17:21.33 | WIMPy | Do yu have a current libsrtp? |
| 17:21.42 | WIMPy | There was a known issue. |
| 17:22.43 | volga629 | <PROTECTED> |
| 17:23.04 | volga629 | yes I put libsrtp let me see which version |
| 17:23.26 | volga629 | Asterisk 1.8.9.2 |
| 17:23.59 | WIMPy | Not Asterisk version. srtp version. |
| 17:24.34 | volga629 | srtp-1.4.2 |
| 17:25.28 | WIMPy | Try 1.4.4 |
| 17:27.44 | volga629 | Ok I am downloading right now see if it helps |
| 17:30.50 | volga629 | dtls_srtp_driver.c:(.text+0x33d): undefined reference to `srtp_profile_get_master_key_length' |
| 17:31.33 | WIMPy | Did you re-make Asterisk? |
| 17:31.49 | volga629 | upgrade ? |
| 17:32.01 | volga629 | yes |
| 17:32.11 | WIMPy | Is that a package? |
| 17:32.41 | volga629 | of libsrtp ? |
| 17:32.47 | WIMPy | Maybe a configute would be a good idea as well. |
| 17:32.57 | WIMPy | No, Asterisk. |
| 17:33.03 | volga629 | no this new 1.4.4 |
| 17:35.42 | volga629 | Ok, Got compiled |
| 17:38.21 | volga629 | ./configure --prefix=/usr CFLAGS=-fPIC |
| 17:38.31 | volga629 | make shared |
| 17:44.04 | tzafrir | coppice, yes |
| 17:44.44 | coppice | tzafrir: someone told me there is a problem with Dahdi 2.6 and OSLEC. Is that right? |
| 17:45.26 | tzafrir | I didn't get to answer that email |
| 17:45.36 | tzafrir | Short version: not that I'm aware of |
| 18:04.12 | volga629 | can't make srtp working properly |
| 18:04.19 | volga629 | with snom 370 |
| 18:06.38 | WIMPy | Does it still crash? |
| 18:09.37 | volga629 | no, I didn't get crashes yes after upgrade, but it trying register on random port and not 5061 |
| 18:09.41 | volga629 | :1029 |
| 18:09.52 | volga629 | I see registration on this port |
| 18:09.57 | volga629 | yet |
| 18:10.38 | WIMPy | Why does that bother you? |
| 18:10.48 | WIMPy | You can switch off randomports, if you like. |
| 18:12.04 | volga629 | Is registration myippbx:randomport instead myippbx:5061 |
| 18:12.08 | volga629 | see |
| 18:12.19 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:12.48 | WIMPy | Sorry, I did not understand that one. |
| 18:14.48 | volga629 | I see in log file 10.234.240.23:1234 port where I set outbound proxy sips:10.234.240.23:5061 |
| 18:16.14 | volga629 | and in asterisk I see registration 1234 random port and not 5061, than mean some issue on transport |
| 18:17.24 | WIMPy | Why do you configure a proxy? |
| 18:17.48 | WIMPy | And are you sure you;ve got the right IP? The random port should be on the phone side. |
| 18:18.19 | *** join/#asterisk Tyrael1 (~Ryan@c-50-129-214-142.hsd1.in.comcast.net) |
| 18:19.26 | volga629 | This phone outside the network I am trying test that is working through outbound proxy |
| 18:19.50 | Tyrael1 | Anybody ever have issues working with Cisco SPA50x series phones not dialing extensions that have wildcards? Such as exten => _*96. ? it works fine for my polycoms the cisco just complains that its either an invalid address or incomplete address. |
| 18:19.56 | volga629 | sips:10.234.240.23:5061 is this registration line should work on client side ? |
| 18:21.18 | WIMPy | I only use user_host = server;transport=tls |
| 18:22.11 | *** join/#asterisk gregor3005 (~Benutzern@h081217007075.dyn.cm.kabsi.at) |
| 18:22.27 | volga629 | let me try |
| 18:26.26 | volga629 | SIP: transport error: 1000000 -> tls:0.0.0.0:5061 |
| 18:26.37 | volga629 | this from snom |
| 18:27.53 | WIMPy | I have no idea how to put a proxy in between. |
| 18:28.33 | WIMPy | Well, "server" was meant to be the name of te server. |
| 18:28.34 | volga629 | no I tried use syntax |
| 18:29.17 | gregor3005 | anybody know where i can find a good documentation for asterisknow, i installed it and found only the freepbx passwort but nowhere the password for the recording section |
| 18:30.34 | ChannelZ | tyman: the dialplan on your phone is messed up |
| 18:31.07 | ChannelZ | sorry not tyman |
| 18:31.12 | ChannelZ | Tyrael1 |
| 18:45.10 | Tyrael1 | how so ChannelZ? |
| 18:45.15 | Tyrael1 | (sorry grabbed a slice of pizza) |
| 18:46.51 | gregor3005 | which sip client is recommended under linux (fedora 16)? |
| 18:49.17 | Tyrael1 | ChannelZ: Or i should say, what should it be? Its the default dial string in the Line1 configuration. I'm not familiar with modifying dial strings at the device level. |
| 18:58.14 | ChannelZ | Well the phone is going to send whatever to Asterisk, either you're not dialing something that matches your extension pattern in Asterisk or the phone is modifying the dialstring before it gets there causing it not to match |
| 18:59.29 | ChannelZ | The example you posted is expecting *96 plus any number of other digits. What is coming from your phone? |
| 19:01.13 | ChannelZ | Actually re-reading what you said it sounds like you probably have IP dialing turned on or something else in the phone |
| 19:01.36 | ChannelZ | (you said it complains of an 'incomplete address') |
| 19:02.22 | WIMPy | Do you think that "address" means IP? |
| 19:03.50 | ChannelZ | It could. I don't have a cisco phone in front of me to see what kinds of things you can break and how it responds. |
| 19:06.21 | Tyrael1 | Well from what you said, i took my limited knowledge of Cisco dial strings and came up with this: (*xxxxx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 19:06.38 | Tyrael1 | that first one there seems to let me dial stuff with a * |
| 19:06.44 | Tyrael1 | before it was just set to *xx |
| 19:07.00 | Tyrael1 | under "Dial Plan" for extension one... which is as you thought. IP Dialing |
| 19:07.09 | Tyrael1 | what happens if i turn that off? |
| 19:07.38 | ChannelZ | you can't dial IP addresses from the phone which you probably weren't doing in the first place |
| 19:08.13 | Tyrael1 | ah.. and correct |
| 19:08.26 | ChannelZ | but chances are it was the first thing.. *xx would only let you dial * and two digits, yet your Asterisk dialplan was probably expecting more |
| 19:08.43 | ChannelZ | without knowing what you're actually doing it's hard to say but in any event.. |
| 19:08.50 | Tyrael1 | I'm also fighting with MWI on the same set of phones... if i have it set to subscribe it just say it has a message. |
| 19:09.01 | Tyrael1 | the Dial Plan change seems to have fixed that specific issue |
| 19:12.40 | ChannelZ | You might actually just be flipping the MWI flag in the phone forcing the light/indication on, not actually setting up MWI |
| 19:13.07 | Tyrael1 | a valid point... man i'm starting to not like these phones... haha |
| 19:14.53 | ChannelZ | do you have mailbox lines in your sip.conf for the devices in question? |
| 19:15.15 | volga629 | I see in log right now No valid transports available, falling back to 'udp'. 'tcp' is not a valid transport type when tcpenabled=no. If no other is specified, the defaults from general will be used. |
| 19:15.49 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 19:18.52 | Tyrael1 | Yes I have the mailbox set up as 98@Context |
| 19:19.42 | ChannelZ | and "Context" is really the name of your voicemail context? |
| 19:19.55 | Tyrael1 | no |
| 19:20.11 | ChannelZ | O.o |
| 19:20.13 | Tyrael1 | it has its own unique name |
| 19:20.22 | Tyrael1 | 98@Zimmerman |
| 19:20.53 | ChannelZ | which matches what 'voicemail show users' shows? |
| 19:21.43 | Tyrael1 | yessir |
| 19:23.49 | ChannelZ | It should then in general just work |
| 19:24.08 | Tyrael1 | I agree.... it should... |
| 19:24.52 | Tyrael1 | do I need to set the Mailbox ID on the cisco or should it just subscribe to the existing connection? |
| 19:24.58 | Tyrael1 | I've tried setting it a few different ways |
| 19:27.30 | ChannelZ | If memory serves Asterisk should offer up to the phone the mailbox it should subscribe to (based on what you set in sip.conf) |
| 19:29.20 | Tyrael1 | i have subscribemwi set to yes |
| 19:30.03 | ChannelZ | where is that? |
| 19:30.30 | Tyrael1 | sip.conf |
| 19:30.35 | Tyrael1 | sibscribemwi=yes |
| 19:30.40 | Tyrael1 | if i could spell subscribe |
| 19:31.16 | ChannelZ | Not sure what it defaults to but I've never in my life explicitly set that |
| 19:31.40 | Tyrael1 | well... i figured it couldnt hurt |
| 19:31.54 | ChannelZ | It probably defaults to yes anyway |
| 19:32.27 | ChannelZ | so if you put a new voicemail in that 98 box the device doesn't notice? |
| 19:33.03 | Tyrael1 | correct. |
| 19:34.04 | ChannelZ | hmm. Not sure specifically for that phone. I don't see any settings in the SPA92x config that jump out. |
| 19:34.19 | Tyrael1 | this one's a SPA509g but should be pretty similar |
| 19:34.34 | ChannelZ | You could turn on SIP debug and reboot the phone and see what they're saying to each other |
| 19:37.24 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:37.31 | ChannelZ | I'm trying to remember how the whole thing even works from the last time I looked at this |
| 19:37.59 | ChannelZ | I don't actually see my phone subscribe to anything, but Asterisk will send a notify to the phone when it has voicemail. |
| 19:40.02 | ChannelZ | yah.. as soon as I leave a voicemail Asterisk sends a NOTIFY to the phone. |
| 19:41.47 | Tyrael1 | http://pastebin.com/f2ShEk3W |
| 19:41.50 | Tyrael1 | thats of the phone reboot |
| 19:45.15 | ChannelZ | hmm |
| 19:46.26 | ChannelZ | it seems to be subscribing to other mailboxes, 703 and 704? |
| 19:46.42 | Tyrael1 | those are park BLFS |
| 19:46.54 | Tyrael1 | and miraculously are working |
| 19:47.37 | Tyrael1 | should be subscribing to 98 (Phones extension), 701-704 (Parks), and BLF For 3 more extensions |
| 19:47.44 | Tyrael1 | all of that seems to be working |
| 19:47.46 | Tyrael1 | less MWI |
| 19:48.15 | ChannelZ | ah ok |
| 19:49.03 | ChannelZ | I'd leave a message and see if you see Asterisk send the NOTIFY to the device |
| 19:49.52 | ChannelZ | Again specifically that phone I don't know, my 922 doesn't even seem to have any config options for subscribing to a mailbox for MWI |
| 19:50.44 | ChannelZ | There's a static switch under Ext1 to turn/show the state of the MWI light... but other than that... hmmm |
| 19:50.59 | Tyrael1 | reading http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf and it says to use that specific light |
| 19:51.06 | Tyrael1 | but that doesn't make sense |
| 19:51.19 | Tyrael1 | "If its not lighting, turn it on all the time" ? |
| 19:53.32 | ChannelZ | hmm it's a pretty weak explanation in that manual. |
| 19:53.42 | ChannelZ | Do you have anything in the Mailbox ID on that Ext tab? |
| 19:53.54 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 19:54.29 | *** join/#asterisk cyborg-one (1000@212-178-11-83.broadband.tenet.odessa.ua) |
| 19:54.48 | Tyrael1 | I've tried just about everything in there |
| 19:55.02 | Tyrael1 | currently have it blank with MWI on, and it leaves the light on no matter what |
| 19:56.29 | ChannelZ | well as I said that literally just turns the light on and off. |
| 19:56.54 | ChannelZ | I would as I said leave a new voicemail and with sip debug still turned on, see if you see Asterisk send the device a NOTIFY about it. |
| 19:57.01 | Tyrael1 | kk one sec |
| 19:57.11 | ChannelZ | And if it does, what the phone replies with. |
| 20:01.16 | ChannelZ | I think maybe it's just out of sync, because it's not a subscribed property but a notification that happens at the time the voicemail is left (or if you go in and delete them.) I just set my MWI flag to off and rebooted and the phone doesn't know there's a VM actually still there because it never asked, and Asterisk never told it. |
| 20:01.36 | Tyrael1 | im using sip set debug peer 95- |
| 20:01.47 | Tyrael1 | and im not seeing anything about it coming back complaining or sending to it |
| 20:01.58 | Tyrael1 | and if i do sip set debug on.. thats a lot of messages to go through |
| 20:02.33 | ChannelZ | your previous paste was for 98- not 95- ? |
| 20:02.44 | Tyrael1 | err 98 |
| 20:03.05 | Tyrael1 | 98 is the phone in question, 95 is another phone, but i'm not playing with that one right now |
| 20:04.24 | ChannelZ | so are you saying that was a type-o and you really were looking at sip debug for 98-whatever? |
| 20:05.33 | Tyrael1 | correct i typed wrong here not in my debug |
| 20:05.42 | ChannelZ | hmm ok |
| 20:06.01 | ChannelZ | So it seems more like * isn't associating the mailbox with that device |
| 20:06.05 | Tyrael1 | http://pastebin.com/hrp9H2St sip set debug on |
| 20:06.12 | Tyrael1 | right before i left a message |
| 20:06.52 | ChannelZ | but what happened right _after_ |
| 20:07.59 | Tyrael1 | it should be in there |
| 20:08.10 | Tyrael1 | i didnt turn it off until about 3 sec after i hung up |
| 20:08.36 | ChannelZ | Here's what I get just after deleting a message I left earlier, but the message is the same (just indicating messages rather than showing 0) http://pastebin.com/PzpsWhFf |
| 20:08.54 | ChannelZ | That's a message sent from * to the device |
| 20:09.29 | Tyrael1 | ok so i should be seeing that from * and am not |
| 20:10.05 | ChannelZ | I can only guess your voicemail/device config is not quite right.. |
| 20:10.35 | ChannelZ | sip show peer 98-30E4DB8077D3 |
| 20:10.52 | volga629 | sip::closed_reg_connection: reregister timer for line 0 set to 61 s what is mean srtp don't want work hmm |
| 20:11.32 | ChannelZ | the Mailbox listed should match the appropriate person in "voicemail show users" |
| 20:11.39 | ChannelZ | Mbox@Context |
| 20:12.23 | Tyrael1 | well voicemail works as far as "Voicemail" alone is concerned. I can log into it and check it, leave it, etc... sip show peer: http://pastebin.com/66VWpPNi |
| 20:13.11 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 20:13.34 | sbrath | has anyone setup a AudioCodes MP-108 w/asterisk? |
| 20:14.13 | Tyrael1 | voicemail show users for Zimmerman = http://pastebin.com/TvWGtkh3 |
| 20:15.28 | ChannelZ | hmmm |
| 20:15.42 | Tyrael1 | i think this box just hates me |
| 20:15.52 | Tyrael1 | as does this phone... |
| 20:19.00 | ChannelZ | Not sure what to suggest. The phone isn't subscribing to MWI events specifically, which isn't necessarily a deal breaker, but I'm not sure why your Asterisk isn't instead sending notifies when the event occurs. |
| 20:19.25 | leifmadsen | do you have polling enabled? |
| 20:19.43 | ChannelZ | My 922 doesn't subscribe either, but Asterisk informs the device when voicemail events on the mailbox occur |
| 20:20.33 | leifmadsen | pollmailboxes=yes |
| 20:20.38 | leifmadsen | pollfreq=XX |
| 20:24.21 | bbourdage | Leif-> yes to both questions, 5 on the freq |
| 20:26.29 | *** join/#asterisk Tyrael1 (~Ryan@c-50-129-214-142.hsd1.in.comcast.net) |
| 20:26.37 | Tyrael1 | sorry about that... gotta love comcast |
| 20:26.47 | Tyrael1 | leif you were saying something ? |
| 20:29.14 | gregor3005 | hi, anybody know wheres the official link to asterisk gui? i found it in asterisknow but i want to install it manually |
| 20:30.08 | leifmadsen | gregor3005: don't think there is one -- to do so manually probably requires you to check out the gui via svn |
| 20:30.28 | gregor3005 | ok, or is it better to use freepbx ? |
| 20:30.37 | leifmadsen | freepbx and asterisk gui are not the same project |
| 20:32.19 | gregor3005 | ok, i tried first asterisknow with freepbx but had some troubles with default password and bad documentation from asterisknow. then i installed the asteriskgui but there are also some troubles that many gui options are not clickable (eg, make backups, configure dialplans, ...) so now i try it manually :-) |
| 20:32.54 | gregor3005 | but the asteriskgui would be nice for my first try in voip |
| 20:33.02 | sbrath | If I recieve a call via a AudioCodes FXO, and the call is routed into asterisk, and I answer. But all I hear is "Loud Audio Scrambled Garbage" is this a Codec problem? |
| 20:33.37 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 20:33.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:48.03 | sbrath | join #audiocodes |
| 20:48.06 | sbrath | oops |
| 20:48.31 | sbrath | Well that dosen't exist anyway. |
| 20:49.20 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 20:57.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:01.21 | *** join/#asterisk dxd828 (~dxd828@88-109-113-33.dynamic.dsl.as9105.com) |
| 21:01.28 | *** part/#asterisk dxd828 (~dxd828@88-109-113-33.dynamic.dsl.as9105.com) |
| 21:02.21 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 21:16.50 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 21:21.46 | ChannelZ | Forget GUIs |
| 21:23.50 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e061.pool.mediaWays.net) |
| 21:26.12 | *** part/#asterisk gregor3005 (~Benutzern@h081217007075.dyn.cm.kabsi.at) |
| 21:31.27 | *** join/#asterisk wonderworld (~ww@dsdf-4db53135.pool.mediaWays.net) |
| 21:34.07 | *** join/#asterisk thx2000 (~thx2000@69-178-135-146.static-ip.telepacific.net) |
| 21:34.57 | thx2000 | If I install IMAP support for asterisk, does that mean I can't use local storage for any of the mailboxes? |
| 21:35.12 | ChannelZ | Tyrael1: I maybe figured out on the Cisco how to get it to specifically subscribe to MWI |
| 21:37.05 | dijib | hello all |
| 21:39.00 | ChannelZ | hallo |
| 21:39.04 | dijib | haro |
| 21:41.03 | sbrath | If I have a Sangoma card with HWEC, does my system.conf still need to specify the mg2 echo canceler? |
| 21:42.34 | sbrath | ok, I guess mg2 is ignored, if HWEC is present. |
| 21:44.07 | sbrath | Occasionally I have a call incoming via Sangoma A400 and while talking you will be silenced for about 6 seconds, then you're back. What would cause this? |
| 21:52.26 | saxa | some kind of hardware conflict maybe |
| 21:52.40 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 21:52.45 | saxa | a dying disk ? :) |
| 21:53.12 | saxa | sbrath: just jokeing |
| 21:53.16 | volga629 | How to change voice mail format from g729 to gsm ? |
| 21:53.41 | saxa | in voicemail.conf ? |
| 21:54.39 | saxa | in [general] add format= |
| 21:54.50 | saxa | not sure if this is the right way |
| 21:55.09 | saxa | i have format=wav |
| 21:57.20 | sbrath | well at least I now people are hearing me now :) |
| 21:57.41 | sbrath | format = wav49|gsm|wav |
| 21:57.49 | sbrath | Will write it out in 3 formats :) |
| 21:57.59 | sbrath | need wav if you want to do email attachments. |
| 21:58.18 | p3nguin | I use wav49 for email attachments. |
| 21:59.34 | volga629 | I have this line format=wav49|gsm|wav |
| 21:59.52 | volga629 | but still prompt in g729 |
| 22:00.19 | volga629 | <PROTECTED> |
| 22:00.37 | p3nguin | If your calls are in g729, asterisk will choose g729 files to play first. |
| 22:01.06 | p3nguin | You asked how to change the voice mail format, not why your prompts are played in g729. |
| 22:02.59 | volga629 | yes, you right I asked because on *97 it play in g729 |
| 22:03.10 | volga629 | it my voice mail |
| 22:03.51 | volga629 | and I changed format option and should not use g729 am I right ? |
| 22:04.08 | p3nguin | No, you are not right. |
| 22:04.53 | p3nguin | Because a prompt file is playing regarding the voice mail system does not make the file your "voiec mail." |
| 22:05.10 | p3nguin | It's just a prompt. It has nothing to do with the recording or playback of your actual voice mail. |
| 22:05.34 | volga629 | I see. just prompt |
| 22:05.44 | ChannelZ | no, just voicemails. |
| 22:06.34 | volga629 | So if user use g729 it will give priority ? |
| 22:06.34 | p3nguin | The format setting in voicemail.conf controls only the format of the recorded voice mails. |
| 22:07.08 | volga629 | AAAAAAAAAAAA I see only recording |
| 22:07.15 | ChannelZ | It will record voicemails in ALL formats listed in voicemail.conf |
| 22:07.15 | p3nguin | Asterisk usually plays the prompts in whatever format matches the codec being used for the call. |
| 22:07.32 | ChannelZ | Which might mean transcoding from the channel format (say, g729) to the recording formats (wav, gsm...) |
| 22:07.45 | volga629 | that why g729, yes all remote use g729 |
| 22:07.53 | p3nguin | If the caller is using ulaw, the prompts will usually be played in ulaw format. |
| 22:07.59 | ChannelZ | so then record your voicemails in g729 (at least) |
| 22:08.06 | p3nguin | If he is using g729, prompts will play in .g729 format. |
| 22:08.27 | p3nguin | If he is using gsm, prompts will play in .gsm format. |
| 22:08.44 | p3nguin | But voice mails will be recorded in the format(s) that you set in voicemail.conf. |
| 22:09.13 | volga629 | you might have some history about this problem ? Unable to open vm-messagex2 (format 0x100 (g729)): No such file or directory |
| 22:10.12 | volga629 | I reopened bug report I so old one in 2009, but was close because reported didn't respond |
| 22:10.20 | sbrath | So has anyone here ever used Audiocodes FXO boards? |
| 22:10.30 | sbrath | I mean devices. |
| 22:10.32 | ChannelZ | it actually says "vm-messagex2" ? |
| 22:10.38 | volga629 | yes |
| 22:11.28 | ChannelZ | oh that's dependent on the language it looks like |
| 22:11.40 | ChannelZ | but anyway it's just a missing sound |
| 22:11.52 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-19431 |
| 22:12.18 | p3nguin | I sure don't have that sound file. |
| 22:12.19 | volga629 | sound file is there I checked |
| 22:12.46 | ChannelZ | show me |
| 22:12.54 | volga629 | I putted link from old bug tracker |
| 22:13.09 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 22:13.37 | learath | anyone use broadvoice? |
| 22:13.41 | ChannelZ | those are two different issues |
| 22:14.17 | volga629 | http://fpaste.org/xOOs/ |
| 22:14.44 | learath | I'm trying to get MWI working w/ broadvoice as a sip provider |
| 22:14.47 | learath | not going well :) |
| 22:15.22 | ChannelZ | volga629: I don't see "vm-messagex2" listed anywhere there |
| 22:15.42 | ChannelZ | or x1 |
| 22:15.46 | *** join/#asterisk nikola_zg (~nix@iskon7120.duo.carnet.hr) |
| 22:16.02 | nikola_zg | hi |
| 22:16.05 | volga629 | yes, file name wrong should vm-message |
| 22:16.07 | volga629 | s |
| 22:16.13 | ChannelZ | not apparently for Russian |
| 22:16.33 | ChannelZ | vm-messagex1 is for "first counting plural form, genative singular" |
| 22:16.46 | ChannelZ | vm-messagex2 is for "second counting plural form, genative plural" |
| 22:16.49 | nikola_zg | i have a question regarding agi and cmd dial |
| 22:17.09 | volga629 | <PROTECTED> |
| 22:17.22 | nikola_zg | I have an AGI which is started when a user calls in. AGI asks for destination number and executes |
| 22:17.22 | nikola_zg | Dial command. After Dial finishes, AGI script catches the sighup to do some cleanup. Is there a way (within the AGI script) to get the billsec variable out of this Dial command? |
| 22:17.38 | nikola_zg | any help appreciated |
| 22:18.12 | volga629 | this line was added from old patch, but it only effected on asterisk older version |
| 22:18.24 | ChannelZ | They're two different issues |
| 22:18.59 | volga629 | I never so file vm-messagex2 |
| 22:19.30 | ChannelZ | All I can tell you is that is the filename it's looking for, and you don't have a sound in any format with that name. |
| 22:19.56 | ChannelZ | The x1 and x2 versions of those sounds are language dependent. Apparently the Russian locale wants to use them, but they aren't in the soundset. |
| 22:20.24 | volga629 | I tried google see might missing file and never can't find nothing about mention file |
| 22:22.29 | ChannelZ | you'll have to record them.. or if the word spoken by "vm-messages" is the same in this instance (it means the same thing) just make a copy with the x2 name |
| 22:23.15 | sbrath | So if I offered $$ to help with getting an AudioCodes working, does anyone know who I could get to help me? |
| 22:23.19 | ChannelZ | (or vm-message as the case may be) |
| 22:24.21 | ChannelZ | sbrath: Is that an FXO-to-SIP device? |
| 22:24.27 | sbrath | yes. |
| 22:24.40 | sbrath | MP-108 |
| 22:25.14 | ChannelZ | and what isn't working? |
| 22:25.18 | volga629 | let me try copy of the file |
| 22:25.21 | sbrath | All I get is whitenoise upon connection to the SIP channel. |
| 22:25.46 | ChannelZ | hmm. fun. |
| 22:25.57 | sbrath | I call, it rings, it routes the call, ulaw 64k, I answer. I'm listening to a bunch of noise.. I'm baffled. |
| 22:26.26 | ChannelZ | You are calling into it or out of it? |
| 22:26.27 | volga629 | when I dial *97 voice mail prompt come to point you don't have new messages and hang the call after that |
| 22:26.43 | sbrath | For now I'm calling the PSTN number and calling into it. |
| 22:26.55 | sbrath | PSTN -> FX108 -> asterisk |
| 22:27.01 | ChannelZ | and you're calling what on the other end? another SIP device? |
| 22:27.56 | sbrath | Hmm.. Call phone -> phone number (PSTN) -> FXO port 1 on MP-108 -> asterisk box -> Yealink T28 phone. |
| 22:28.03 | sbrath | s/call phone/cell phone |
| 22:28.20 | ChannelZ | So the noise is the incoming audio from the Audiocodes box.. does your outgoing audio do the same? |
| 22:28.40 | sbrath | I have not tried to use it outgoing, I can try it. |
| 22:28.44 | ChannelZ | no |
| 22:28.55 | sbrath | I hear the noise on the Cell phone as well as the yealink phone. It's really loud. |
| 22:28.57 | ChannelZ | I mean two-way audio |
| 22:29.06 | ChannelZ | so it's noise in both directions |
| 22:29.08 | sbrath | yes |
| 22:29.38 | ChannelZ | Hmm. Can you configure different codecs on the MP108 box? I wonder if it's broken in the head. |
| 22:29.59 | ChannelZ | sending bogus RTP data.. or has ulaw vs alaw confused or somesuch |
| 22:30.01 | sbrath | I tried G729 but couldn't transcode it on the asterisk, alaw was the same. |
| 22:30.13 | ChannelZ | wierd. |
| 22:30.17 | sbrath | I tried alaw as the primmiary asterisk freaked. |
| 22:30.30 | ChannelZ | what do you mean |
| 22:30.33 | sbrath | ulaw64k is the default. |
| 22:30.56 | ChrisInSydney | sbrath: You in the US ? ulaw is default |
| 22:31.01 | volga629 | I copied file it not failing right now, but it not detecting new message |
| 22:31.03 | sbrath | asterisk couldn't figure out how to transcode the alaw that it was sending to something my phone would understand, |
| 22:31.08 | sbrath | I'm in the US |
| 22:31.16 | sbrath | Wisconsin in fact . |
| 22:31.28 | ChrisInSydney | K then set ulaw, to make life easier |
| 22:31.35 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 22:31.43 | mattwj2002 | hi guys |
| 22:31.47 | sbrath | Yes, I have it at ulaw, still sounds like crap. |
| 22:32.03 | ChrisInSydney | Just a had aquick read of your issues. I had a similar issue with a SPA3102 |
| 22:32.45 | ChrisInSydney | what settings do you have to set up the impedance / capacitance characteristics on the FXO line? |
| 22:32.54 | sbrath | let me check. |
| 22:33.11 | ChrisInSydney | did you check that, make suire its not setup for another continent |
| 22:33.21 | ChrisInSydney | justa small thing |
| 22:33.29 | volga629 | <ChannelZ>: What can cause not detect messages in voice mail box ? |
| 22:33.53 | ChannelZ | volga629: what do you mean "not detecting"? |
| 22:34.03 | ChrisInSydney | volga629: you using realtime for VM ? |
| 22:34.10 | ChannelZ | You were messing around with the recording formats earlier, probably your mailboxes are all out of sync |
| 22:34.40 | mattwj2002 | I have a question is there a way to search which resellers have which numbers? |
| 22:34.59 | volga629 | I left voice mail on my snom test phone and when I dial *97 it says no new voice messages |
| 22:35.05 | mattwj2002 | finding local rural wisconsin numbers are difficult |
| 22:36.38 | ChannelZ | Is *97 accessing the same mailbox you're leaving messages into? (is this FreePBX?) |
| 22:37.42 | volga629 | Yes |
| 22:37.59 | volga629 | I see in log when recording |
| 22:38.02 | volga629 | <PROTECTED> |
| 22:38.02 | volga629 | <PROTECTED> |
| 22:38.03 | volga629 | <PROTECTED> |
| 22:38.46 | volga629 | might be some thing on snom phone him self ? |
| 22:39.25 | ChrisInSydney | volga629: Are you using the snom voicemail button, or are you dialling *97 ? |
| 22:39.42 | ChrisInSydney | what happens on the CLI when you call *97 ? |
| 22:39.59 | ChrisInSydney | Should be checking 101 from your previous post |
| 22:40.34 | volga629 | yes voice mail button |
| 22:41.06 | volga629 | let me paste bin output from *97 |
| 22:41.17 | ChrisInSydney | what happens when you just dial *97 ? |
| 22:42.11 | mattwj2002 | hey any help? |
| 22:42.27 | volga629 | http://fpaste.org/ekcu/ |
| 22:42.33 | volga629 | that what I see |
| 22:42.39 | ChrisInSydney | Uness you have programmed the VM button as speeddial *97 as opposed to a key event |
| 22:43.31 | ChannelZ | as I said your mailbox is probably all screwed up |
| 22:43.40 | volga629 | I tried dial from voice mail button and from dial pad same result |
| 22:43.57 | ChannelZ | If you don't have any important saved messages, just wipe out the whole mailbox directory |
| 22:44.12 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 22:44.26 | volga629 | No this test environment only |
| 22:44.30 | ChannelZ | or at least your inbox |
| 22:44.36 | ChrisInSydney | damn I need a new IRC client |
| 22:44.42 | ChannelZ | /var/spool/asterisk/voicemail/default/101/INBOX |
| 22:45.15 | volga629 | it empty |
| 22:45.30 | ChannelZ | hmm. then what mailbox were you leaving messages into? |
| 22:45.46 | ChannelZ | oh nevermind you showed 101 above |
| 22:45.56 | ChrisInSydney | volga629: CLI: show voicemail users |
| 22:46.13 | ChannelZ | Were they really saving? (leaving long enough messages?) |
| 22:46.30 | ChannelZ | Did the console spit any warnings after you left the message? |
| 22:46.43 | ChrisInSydney | volga629: disk space ? permissions ? |
| 22:46.50 | volga629 | no |
| 22:47.34 | *** part/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 22:48.10 | volga629 | let me check again |
| 22:48.20 | ChrisInSydney | mattwj2002: Not really. Just google, sorry. I'm in Au so I cant even recommend |
| 22:48.48 | ChrisInSydney | sbrath: So how are things going ?? |
| 22:49.45 | volga629 | No such command 'show voicemail users' |
| 22:49.53 | ChannelZ | it's "voicemail show users" |
| 22:50.14 | ChrisInSydney | ChannelZ: Still stuck in Cisco land :-/ |
| 22:50.27 | ChrisInSydney | or am i just dyslexic |
| 22:50.37 | ChannelZ | heh |
| 22:50.37 | volga629 | new vm all 0 |
| 22:50.51 | ChrisInSydney | maybe it was too much time with the hookers 8p |
| 22:51.21 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 22:51.29 | ChrisInSydney | volga629: thats why we dont use FreePBX here |
| 22:51.31 | volga629 | it not permissions or space owner and group asterisk:asterisk |
| 22:51.59 | ChannelZ | then be sure your minlength for voicemails isn't like 5 seconds and you're only talking for 3 |
| 22:52.46 | volga629 | I remember I so somethink line 3m in voicemail.conf |
| 22:52.55 | volga629 | sorry 3sec |
| 22:53.44 | volga629 | that line in voicemail.conf minsecs=3 |
| 22:54.51 | ChannelZ | and are you talking for more than 3 seconds? not silence? |
| 22:55.05 | ChannelZ | (or maybe your audio doesn't work and you just don't know it yet? :) |
| 22:55.11 | volga629 | no talking about 15 20 sec |
| 22:55.21 | volga629 | :-) |
| 22:55.32 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 22:55.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:55.42 | volga629 | than interesting point |
| 22:55.43 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:56.00 | ChannelZ | well the console with a little verbose on should reveal what is happening to the message after you leave it |
| 22:56.51 | volga629 | Parsing '/var/spool/asterisk/voicemail/default/101/INBOX/msg0000.txt': == Found |
| 22:57.44 | volga629 | and this next line |
| 22:57.47 | volga629 | Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/babyTel-00000014' in macro 'vm' |
| 22:59.13 | volga629 | If I am understand write it not completing with message was recorded |
| 22:59.20 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 23:00.11 | ChannelZ | well it seems like it is but you're saying that after leaving a message, /var/spool/asterisk/voicemail/default/101/INBOX is empty?? |
| 23:01.20 | ChrisInSydney | volga629: I sould simply script something up to directly call voicemail followed by voicemailmain |
| 23:01.27 | ChrisInSydney | test that |
| 23:02.13 | thx2000 | volga629: what does your voicemail.conf look like for that extension? |
| 23:03.16 | volga629 | yes empty |
| 23:03.34 | ChannelZ | errr |
| 23:03.35 | ChrisInSydney | volga629: Does the MWI light show on the Snom ?? |
| 23:03.45 | volga629 | yes |
| 23:03.55 | volga629 | is light MWI on |
| 23:07.19 | volga629 | http://fpaste.org/4Bez/ |
| 23:07.26 | volga629 | this voicemail.conf |
| 23:07.29 | p3nguin | /var/spool/asterisk/voicemail/default/101/INBOX must have wrong permissions. |
| 23:08.28 | volga629 | http://fpaste.org/aXzG/ |
| 23:08.34 | volga629 | this permissions |
| 23:08.37 | p3nguin | If the message file cannot be written, asterisk might indicate that it was when it really isn't. |
| 23:09.14 | volga629 | i has 755 on folder |
| 23:09.18 | volga629 | it |
| 23:09.20 | ChrisInSydney | must got to a service call. Damn Exlastix system. :-/ |
| 23:09.23 | ChrisInSydney | c yaz |
| 23:09.24 | p3nguin | I don't fold things. |
| 23:09.48 | ChrisInSydney | volga629: Good luck, you are in capable hands |
| 23:10.15 | volga629 | Thank you for you help and time <ChrisInSydney> |
| 23:11.03 | ChannelZ | I assume you deleted the mailboxes from your paste of voicemail.conf and that they really are there? |
| 23:11.38 | *** join/#asterisk Abn0rmal (~jack@2001:1938:155:0:52e5:49ff:fec4:e7df) |
| 23:12.07 | ChannelZ | freaking FreePBX |
| 23:12.08 | ChannelZ | go ask them |
| 23:12.12 | thx2000 | Yeah, I'd like to see the actual mailbox config in voicemail.conf. Should be under '[default]' |
| 23:12.32 | ChannelZ | convoluted piece of trash |
| 23:12.34 | Abn0rmal | Can Asterisk be used to create an internet call in show (similar to Blogtalk Radio)? |
| 23:12.51 | volga629 | <PROTECTED> |
| 23:13.38 | thx2000 | volga629: cat /etc/asterisk/voicemail.conf |
| 23:14.09 | volga629 | or sorry wrong one, yes just sec |
| 23:14.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:15.35 | volga629 | http://fpaste.org/waBL/ |
| 23:16.13 | thx2000 | volga629: 101 => XXXX,test,test@networklab.ca,,attach=yes|saycid=yes|envelope=yes|***delete=yes*** |
| 23:16.21 | thx2000 | volga629: delete=yes is your problem |
| 23:16.54 | thx2000 | volga629: It's attempting to send an e-mail to test@networklab.ca then wiping the message from local storage |
| 23:18.03 | ChannelZ | heh nice |
| 23:18.09 | volga629 | let me remove delete |
| 23:21.28 | volga629 | -- Executing [s-NOANSWER@macro-vm:3] Goto("SIP/babyTel-00000019", "exit-SUCCESS,1") in new stack |
| 23:23.11 | volga629 | yes it working right, but prompt in russian still broken, but this is another story |
| 23:23.21 | thx2000 | :) |
| 23:24.01 | volga629 | Everyone thank you for you help and time this good lesson |
| 23:24.07 | ChannelZ | maybe Russian is broken |
| 23:24.44 | volga629 | is russian prompt will be fixed or need to do our self ? |
| 23:25.04 | ChannelZ | it's apparently missing from the sound set |
| 23:25.08 | ChannelZ | why I don't know |
| 23:25.21 | volga629 | yes it is broken Unable to open digits/1n (format 0x100 (g729)): No such file or directory |
| 23:26.08 | ChannelZ | 1n is "neuter singular for phrases such as "one message" or "thirty one messages"" |
| 23:28.57 | volga629 | yes I see digit directory under ru |
| 23:29.57 | volga629 | and I see something like this 1.g729 |
| 23:30.15 | ChannelZ | yes but 1n is a separate sound |
| 23:30.42 | volga629 | yes might be this part missing need check |
| 23:30.48 | *** part/#asterisk Abn0rmal (~jack@2001:1938:155:0:52e5:49ff:fec4:e7df) |
| 23:31.06 | ChannelZ | from app_voicemail: http://pastebin.com/2AYVzBe6 |
| 23:32.25 | ChannelZ | Why these additional sounds aren't in the standard russian sound set, I don't know. |
| 23:34.03 | volga629 | so need look might be some patch for this |
| 23:34.41 | ChannelZ | well someone needs to record those additional words |
| 23:34.55 | ChannelZ | Is the Russian soundset Allison? |
| 23:35.35 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:38.55 | sawgood | outside of an incorrect password, what else would return a 401 "unauthorized" message? |
| 23:40.14 | ChannelZ | A command issued by someone who didn't authenticate at all and needs to |
| 23:40.57 | volga629 | <ChannelZ: Yes I will look for those files, might be need pay for normal quality recording |
| 23:41.07 | ChannelZ | It's actually sort of normal.. a device tries to register, gets an Unathorized response which also includes a hash which the device re-registers with auth using. |
| 23:41.21 | sawgood | thats right ... I've seen that many times ... |
| 23:41.36 | sawgood | thats why it does that |
| 23:43.27 | sawgood | I have an * 1.8.9.2 box trying to register to an ITSP with a registration string ... (it worked before switching from DSL to cable) ... now sip show registry is 'stuck' with 120 "request sent" |
| 23:44.03 | sawgood | The sip messages are arriving to the ITSP ... but no registration is happening |
| 23:44.20 | volga629 | more like NAT issue |
| 23:44.32 | ChannelZ | how do you know they're getting there? do you see responses in a sip debug? |
| 23:44.51 | sawgood | I know they are getting the messages because they tell me |
| 23:45.28 | ChannelZ | Are you getting 401s back from them? Or nothing? |
| 23:46.15 | sawgood | yes 401 messages come back to the * box from the ITSP |
| 23:47.42 | ChannelZ | So they don't like your auth for some reason you'd have to ask them. Were you on a static IP before and they knew that (in which case you wouldn't really have to register in the first place?) |
| 23:47.50 | ChannelZ | You said you switched ISPs so... |
| 23:47.57 | *** join/#asterisk cyborg-one (1000@212-178-2-177.broadband.tenet.odessa.ua) |
| 23:48.05 | sawgood | actually, the messages on the * box are so fastly displaying ... they might not be return messages |
| 23:48.27 | sawgood | I have debug on for the ITSP address ... but it probably is showing outgoing messages as well |
| 23:48.54 | sawgood | static IP before and DHCP on WAN side now |
| 23:49.03 | sawgood | the ITSP has host=dynamic |
| 23:49.14 | ChannelZ | I would guess you're NOT getting responses based on the fact your thing just says 'request sent' but I might be wrong |
| 23:49.27 | ChannelZ | Is your * box behind a firewall? |
| 23:49.42 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 23:49.44 | sawgood | no firewall (DD-WRT router wide open for testing) |
| 23:49.51 | sawgood | ports forwarded to * from DD-WRT |
| 23:50.30 | sawgood | remotely, I can SSH to the * box, so port forwarding is working for sure |
| 23:51.07 | ChannelZ | hard to say without knowing if you're getting 401s or some other response, or not getting any response at all. |
| 23:51.23 | sawgood | I think the local * box is not getting anything at all |
| 23:51.40 | ChannelZ | me too |
| 23:52.04 | sawgood | The ITSP side gets (2) SIP messages over and over (register) and then (unauthorized) |
| 23:52.35 | ChannelZ | hmmm why would THEY get an authorized? |
| 23:52.59 | sawgood | I can paste bin what shows up at the ITSP if you would look at it? |
| 23:53.18 | ChannelZ | yeah worth a peek |
| 23:53.26 | ChannelZ | I'm confused about what is really transpiring |
| 23:55.52 | sawgood | http://pastebin.com/hpAyhZ9W |
| 23:55.58 | sawgood | That is the ITSP side |
| 23:56.04 | sawgood | I'll give you the client side shortly |
| 23:57.16 | sawgood | http://pastebin.com/UFkTvgcL |
| 23:57.18 | sawgood | client side |
| 23:57.19 | ChannelZ | ok the 401s are what they are sending you back |
| 23:57.45 | sawgood | I guess those are not arring back to the client? |
| 23:57.47 | ChannelZ | Trying to get you to auth. But your side is not getting the responses so after a time it just tries sending the REGISTER again. |
| 23:58.12 | sawgood | cool ... what causes a break down like that? |
| 23:58.22 | ChannelZ | ?? |
| 23:58.41 | sawgood | I guess I can use dyndns.org to solve this for now? |
| 23:58.43 | ChannelZ | Those packets are getting stopped by your cable modem or router or firewall, probably |
| 23:58.55 | sawgood | new cable modem router (Motorola) |
| 23:58.58 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 23:58.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:59.05 | sawgood | thank you for your time/help! |
| 23:59.07 | ChannelZ | Comcast? |
| 23:59.18 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 23:59.20 | sawgood | Yes, but here is a local city owned version of Comcast cable |
| 23:59.27 | sawgood | The 'city' is the ISP |
| 23:59.48 | sawgood | We are going to get a static IP tomorrow for sure! |
