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| 00:24.10 | simprix | Anyone have experience using a cisco router as a voice gateway? |
| 00:25.37 | sawgood | like in "Call Manager"? |
| 00:25.56 | simprix | no just using a router to terminate a pri |
| 00:26.18 | sawgood | Is the PRI in the router? |
| 00:26.34 | simprix | yes |
| 00:26.53 | sawgood | So, nothing to do with an Asterisk box behind the Cisco router? |
| 00:27.27 | simprix | Then we have a sip dial peer to a asterisk pbx. I'm just looking for a debug command to show that the ani is being presented to the pri correctly for the carrier |
| 00:42.55 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 00:44.43 | WIMPy | "Just a router" does SIP? |
| 00:45.19 | WIMPy | 'sip set debug peer xxx' |
| 00:58.31 | carrar | Cisco router with some DSP action |
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| 01:31.03 | banditti | I have a cisco 7940 that just says "configuring IP" and I can't get into settings, or anything at all. Thoughts? |
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| 01:33.36 | banditti | please? |
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| 02:22.52 | tyman | is the extensions.conf the only config file that supports the 'include =>' directive? |
| 02:24.16 | russellb | yes, that is extensions.conf specific |
| 02:24.27 | russellb | however, all files support #include "filename" |
| 02:24.43 | tyman | oh...cool, that works |
| 02:24.47 | tyman | thanks... |
| 02:24.48 | russellb | and all files support configuration section inheritance |
| 02:25.30 | tyman | hmmm....is that covered in your book? |
| 02:25.59 | russellb | should be ... i'm looking |
| 02:27.47 | russellb | i can't find it covered specifically, but here is one example at least: http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceStates.html#SLA |
| 02:27.57 | russellb | look for ... [station](!) |
| 02:28.00 | russellb | that's a template |
| 02:28.09 | tyman | oh...templates...got it |
| 02:28.09 | russellb | [station1](station) |
| 02:28.16 | russellb | that's a section that inherits from a template |
| 02:28.22 | tyman | y...i know about that. gotcha now |
| 02:28.26 | russellb | k |
| 02:29.02 | tyman | saw that in the book too...i just mis-understood.. thanks for your help |
| 02:29.55 | russellb | np |
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| 02:43.37 | Kobaz | do de do |
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| 03:17.59 | kaushal | Hi |
| 03:18.45 | kaushal | Any clue about http://pastebin.ubuntu.com/636150/ ? |
| 03:18.55 | kaushal | I am on CentOS 5.6 |
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| 03:45.30 | carpe_diem | hey |
| 03:45.55 | carpe_diem | can someone explain me the rationale behind asterisk NOT supporting jackd? |
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| 03:50.31 | drmessano | carpe_diem: Since when? |
| 03:56.36 | carpe_diem | all I see is alsa useflag in the gentoo ebuild and not jack. |
| 03:57.28 | carpe_diem | and oss |
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| 04:34.39 | kaushal | checking in again for the query ? |
| 04:38.40 | linuxgecko | ?? |
| 04:39.46 | kaushal | Any clue about http://pastebin.ubuntu.com/636150/ ? I am on CentOS 5.6 |
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| 04:43.44 | linuxgecko | kaushal: looks like a permission issue, specific to ubuntu.. tried asking the ubuntu guys? |
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| 04:45.21 | linuxgecko | anyone here that might be able to help me tweak the settings on my RTP300 so it will properrly register with my asterisk? i don't need it unlocked.. i already did that :) |
| 04:45.46 | drmessano | Nothing to tweak. From default, just need a proxy, user, and password |
| 04:46.27 | drmessano | I forgot I even had a few of those |
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| 05:07.31 | tyman | It appears a 'dialplan show/reload' doesn't expand the include'd files on the console...is there a workaround for this? |
| 05:09.04 | tyman | clarification: it shows that they're included, it doesn't display the active dialplan entries the files contain using those commands. I must be missing something |
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| 05:47.06 | kaldemar | tyman: it does. pasteib what you're doing exactly. |
| 05:47.37 | kaldemar | s/pasteib/pastebin |
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| 05:49.33 | the_tux | hello everyone |
| 05:49.40 | the_tux | need little help with asterisk |
| 05:49.59 | the_tux | facing sound-breakage while calling.. any suggestions to overcome this issue? |
| 05:50.22 | tyman | kaldemar: using #include "filename" works as expected. include => ... puts the include'd directives under the last [context] read in the extensions.conf |
| 05:51.23 | tyman | even when there are [context] statements in the file included by include => |
| 05:51.36 | tyman | i see how it works now... |
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| 06:11.26 | the_tux | facing sound-breakage while calling.. any suggestions? |
| 06:11.50 | jsjc | I am getting this   -- Got SIP response 480 "Temporarily Unavailable" back from |
| 06:11.50 | jsjc | 1:04 |
| 06:11.50 | jsjc | and do not kow why because my setup in a softphone is the same |
| 06:11.50 | jsjc | 1:05 |
| 06:11.50 | jsjc | it is registered because sip show peers shows connected... |
| 06:13.59 | kaldemar | tyman: include => does not include files, just contexts. |
| 06:14.47 | kaldemar | jsjc: a DND mode on in the phone? |
| 06:14.51 | tyman | kaldemar: i just read the section in the * def guide...got it...i was being an idiot |
| 06:15.20 | jsjc | using a sofphone to connect... |
| 06:15.32 | jsjc | no dnd on the phone. |
| 06:16.08 | jsjc | if i connect to the sip trunk direct with sift phone I can make calls. |
| 06:16.14 | jsjc | When i connect my asterisk to the trunk |
| 06:16.19 | jsjc | i canno make the calls |
| 06:17.51 | the_tux | is anyone knows how to troubleshoot sound-breakage issue? I am not able to talk at all as sound is disconnects frequently :( |
| 06:20.36 | tyman | i'm getting my ears blown the F*** out when I hear congestion thru my softphone. Question: is the congestion volume/tone from the indications.conf or from the client? |
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| 06:29.15 | tonsofpcs | congestion? |
| 06:31.39 | linuxgecko | anyone here good with rtp300's? i know i posted this a while ago, but my comp crashed, and i didn't have a log setup. i keel getting 401 unautorized and 403 bad auth from sip debug. i coppied the info stright from sip.conf, and this same info works for sipdroid , and also ekiga. my asterisk is 1.8.4.2. |
| 06:31.55 | linuxgecko | s/keel/keep/ |
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| 06:38.03 | tyman | tonsofpcs: yes...congestion notification |
| 06:40.47 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:40.49 | schmidts | good morning |
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| 07:00.28 | kleszcz | morning |
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| 07:17.56 | lyetz | Can anybody recommend a Philippines (+63) DID provider? |
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| 08:04.48 | irroot | new t38 gateway patch coming up soon backporting the cleanups mnicholson has put in looking really good thx for the hard work |
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| 09:01.14 | Mezevenf | hello, I'm running a B410P on Telstra ISDN but I cannot seem to recieve calls. Outbound works fine however. Calls do not seem to reach the system |
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| 09:08.57 | banditti | Question. For dial by name, is there a way to do that multi tenant? If I had several customers on my switch each could have one? |
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| 09:23.17 | tzafrir_laptop | banditti, what do you mean? |
| 09:23.42 | tzafrir_laptop | DAHDI dial by name? |
| 09:24.16 | banditti | Say I have company A and company B. I want two different pools of names to dial from. Can you do that or does it pick from all extension on the switch? |
| 09:24.53 | tzafrir_laptop | banditti, that's a matter of the dialplan |
| 09:25.00 | tzafrir_laptop | The namespace is global |
| 09:25.32 | tzafrir_laptop | I can't think of a way to tell one tenant from another to the DAHDI kernel |
| 09:25.55 | banditti | That is what I was afraid of. |
| 09:26.03 | tzafrir_laptop | But if you manage to restrict them in the dialplan or whatever to a single pool, it should work |
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| 09:29.22 | war9407 | Anyone here use hylafax server ? |
| 09:29.39 | irroot | uses hyla |
| 09:29.41 | banditti | that would just restrict comp a from seeing comp b |
| 09:29.55 | banditti | it wouldn't hold true vise versa would it? |
| 09:29.58 | war9407 | loves hylafax, was curious do you have an smtp(POSTFIX) to hylafax gateway? |
| 09:30.52 | irroot | dont use postfix |
| 09:30.56 | war9407 | oh ok |
| 09:31.03 | war9407 | but do you have an SMTP gateway? |
| 09:31.05 | war9407 | for Faxes |
| 09:31.06 | irroot | and dont trust mail -> fax |
| 09:31.09 | war9407 | k |
| 09:31.25 | irroot | do have fax->mail sendmail is my MTA |
| 09:31.38 | war9407 | yep, I do as well, seems to work good -> to PDF |
| 09:32.03 | irroot | i use * for fax->mail though not hyla |
| 09:32.13 | war9407 | oh |
| 09:32.19 | irroot | hyla is outbound "print too fax" |
| 09:32.34 | war9407 | ah, thats even better actually |
| 09:32.41 | war9407 | how did you set that up? |
| 09:32.46 | war9407 | samba -> hylafx -> ? |
| 09:33.15 | irroot | using a windows hylafax print driver not via samba |
| 09:33.20 | war9407 | ahh ok |
| 09:33.25 | war9407 | still you gave me a good idea, thanks |
| 09:33.39 | irroot | t38modem v.2 -> gnugk -> ooh323 |
| 09:33.55 | irroot | with T.38 gateway it goes via TDM lines |
| 09:34.27 | war9407 | awesome |
| 09:34.29 | war9407 | mine is: |
| 09:34.52 | war9407 | Goole voice (free) -> Obi110($40-50/USD) -> USB Modem -> hylafax |
| 09:38.30 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 09:38.35 | jacc0 | hi all :) |
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| 11:56.09 | jacc0 | why is it so quiet in here? puplic holiday somewhere? |
| 11:56.24 | jacc0 | or did they fix all the bugs in asterisk :P |
| 11:56.27 | irroot | not here but its a friday |
| 11:56.31 | irroot | :P |
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| 12:08.41 | vragnaroda | it's a friday preceding a long weekend here. probably a lot of people have today off, too |
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| 12:29.03 | bobb_WU | would someone talk to me about reinvites, native bridging vs. p2p, and how to make sure my relay is working like i want it to? |
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| 12:45.12 | bobb_WU | is anybody around? |
| 12:45.59 | Sylnai | try asking a specific question, it might get a better response |
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| 12:57.24 | bobb_WU | ok. how do i verify that my relay server is passing off revinvites so that its not wasting processor power? |
| 12:59.49 | bobb_WU | i have some pretty specific questions about what i'm confused about. please someone just start a convo about it! |
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| 13:10.31 | WIMPy | Unless you are transcoding you don't need to worry about CPU load. |
| 13:11.51 | bobb_WU | that's not the question though |
| 13:12.13 | bobb_WU | its a virtual machine in a cluster so really i don't have to worry about it to start with |
| 13:12.58 | bobb_WU | but all my nodes are on the same switch, so i want the relay to do the SIP traffic then pass off the RTP to just being between the nodes |
| 13:13.33 | bobb_WU | no need for the relay to touch all the voice traffic on our system |
| 13:14.25 | bobb_WU | on voip-info.org, i saw that canreinvite (1.4) became directmedia (1.6) but i've tried both and still see native bridging on the relay |
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| 13:15.28 | WIMPy | Do you have any features enabled that require Asterisk to stay in the media path? I.e. anything that is controlled via DTMF? |
| 13:17.40 | bobb_WU | nothing special is enabled. we use the relay to defeat the licensing issue of only 15 concurrent connections to the propreitary VOIP system on campus |
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| 13:18.31 | WIMPy | That somehow sounds more like a yes. |
| 13:18.32 | bobb_WU | so the logic of the relay is simple: set a redirect flag, add a sip header, dial (forward) |
| 13:18.41 | rcaskey | hello all, I'm in the atlanta metro wand was looking for suggestions on who to do sip trunking with? |
| 13:18.50 | WIMPy | Dynamic features do require DTMF. |
| 13:19.10 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 13:19.21 | bobb_WU | well... the dynamic part comes from pulling the extension data from the database |
| 13:20.00 | WIMPy | No. "dynamic features" like transfer. |
| 13:20.00 | bobb_WU | and we are getting peers and reg data from the db as well, though we have no SIP clients (only analog phones connected to their own Asterisk boxes) |
| 13:21.45 | bobb_WU | there's no user/phone accessible logic on the relay. it should simply set up connections and thereafter, let the end points talk directly |
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| 14:19.11 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124) |
| 14:19.40 | eduzimrs | anyone know about REDFONE FONE BRIDGE?? |
| 14:21.23 | eduzimrs | im running tcpdump in its interface and so, print a PACKAGE EXPLOSION ... |
| 14:22.19 | _Corey_ | eduzimrs: Redfone uses TDMoE and consumes a solid amount of bandwidth on its interface |
| 14:23.03 | *** join/#asterisk jacobkiers (~jacobkier@82-168-215-74.ip.telfort.nl) |
| 14:23.32 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:31.45 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:33.31 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
| 14:34.12 | RickB17 | anyone here that can help me debug a crashing asterisk service? i'm familiar with the console, but not debugging. |
| 14:34.29 | RickB17 | if you can point me to a doc or something that would be very helpful. |
| 14:34.55 | irroot | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:35.50 | RickB17 | thanks |
| 14:35.53 | RickB17 | i'll take a look |
| 14:36.04 | irroot | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 14:36.35 | irroot | RickB17 depends whats happening maybe mention the problem someone may have it at hand |
| 14:36.49 | RickB17 | when i call one extension my service crashes |
| 14:36.55 | RickB17 | but every other extention works fine |
| 14:37.10 | RickB17 | to dial an ext i use a macro and they all use the same macro |
| 14:37.14 | RickB17 | so thats where i'm lost |
| 14:37.32 | irroot | well get a dialplan trace pastebin it and lets see |
| 14:37.37 | RickB17 | k |
| 14:37.39 | irroot | im off home soon but othere are here |
| 14:39.40 | RickB17 | output from the console: http://pastebin.com/6dwFMbrw |
| 14:42.02 | irroot | put your dialplan up as well please |
| 14:42.22 | RickB17 | k |
| 14:42.32 | RickB17 | i am using the 1.8.5 rc1 |
| 14:42.45 | RickB17 | i might try to roll back to see if it's a new issue |
| 14:42.51 | RickB17 | but i'll try to work through this first |
| 14:43.43 | RickB17 | dial plan: (well the macro) http://pastebin.com/xwNijkJ3 |
| 14:44.00 | RickB17 | if you see anythign that i am doing that might be "wrong" or you know of a better way, let me know. |
| 14:44.46 | irroot | internal-in ?? thats where it fails |
| 14:45.09 | RickB17 | k one second |
| 14:45.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:45.56 | RickB17 | internal-in: http://pastebin.com/ZjKF9ZHX |
| 14:46.34 | *** join/#asterisk sarelon (~sarelon@24-116-133-114.cpe.cableone.net) |
| 14:47.07 | irroot | mmm get a backtrace please |
| 14:48.13 | RickB17 | k |
| 14:48.16 | RickB17 | i'll give that a go |
| 14:49.08 | RickB17 | thanks |
| 14:52.17 | *** join/#asterisk Guizmo (~Guizmo@unaffiliated/guizmo) |
| 14:54.29 | Guizmo | hello, I am getting a problem with asterisk for an outbound route. I always get: Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0 (everyone is busy/congested at this time (1:0/0/1). How can I find the main cause ? |
| 14:55.12 | eduzimrs | anyone know about REDFONE FONE BRIDGE?? |
| 14:56.04 | ruyo | I did a tcpdump and realized most of the calls had the RTP packet with "marker" as SET (usually the first, I think) with status either "Wrong sequence nr." or "Incerrect timestamp". What can cause this? What can this cause? |
| 14:56.47 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:57.43 | ruyo | Guizmo, if you call, for instance, a SIP phone that's disconnected, you'll get that response, I think. |
| 14:58.08 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:58.08 | Guizmo | I am in PSTN only, I use only sip locally. I can phone sip phone, but have problem with the outbound route for the PSTN |
| 14:58.53 | Guizmo | (have a Digium B410P card for an ISDN BRI line) |
| 14:59.30 | ruyo | mISDNv1 or v2? |
| 15:01.58 | Guizmo | is it the driver ? i am using dahdi |
| 15:02.20 | ruyo | Ah. |
| 15:02.43 | ruyo | I never used dahdi for ISDN. |
| 15:02.50 | ruyo | Can you receive calls? |
| 15:03.39 | Guizmo | didn't try yet, was trying to get the outbound working :) |
| 15:04.09 | WIMPy | Guizmo: Try the other way. There have been known issues with L1 (de)activation. |
| 15:04.40 | WIMPy | Or maybe still are. I've never tried to debug that part. |
| 15:04.42 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:09.06 | ruyo | I never did have problems with it, but just in case I started using misdn_check_l2l1(). |
| 15:09.47 | WIMPy | That's only for chan_misdn. |
| 15:10.56 | ruyo | Yeah. |
| 15:13.07 | linuxgecko | anyone here good at connecting an rtp300 ata to asterisk? all google can seem to tell me is tat people have trouble unlocking them from vonage, which i seem to have already done. but i keep getting authentication issues, according to sip debug on my 1.8.4.2 asterisk console.. ideas? |
| 15:16.18 | *** join/#asterisk _omer (~omer@182.185.148.245) |
| 15:16.19 | Guizmo | I cannot seem to call my softphone from an other line. Asterisk doesn't even log the incoming call |
| 15:16.57 | WIMPy | Guizmo: Has anything worked before? |
| 15:17.59 | Guizmo | it is the first time I try to connect sip to the PSTN. The P1 of the card is green thought |
| 15:19.04 | WIMPy | 'dahdi show status' and 'dahdi show channels' |
| 15:20.23 | _omer | I have a general question about Vicidial, I read that it is based on Asterisk so it should produce Manager events exactly like asterisk .... correct? |
| 15:23.35 | irroot | _omer vicidial is last i looked at it asterisk 1.2 was what it used this is not supported but should be similar |
| 15:24.13 | Guizmo | WIMPy: http://pastebin.com/mzd3G4S8 |
| 15:24.19 | sarelon | anyone familiar with freepbx 2.9.0.7 - need some help changing the login uid/pass |
| 15:24.33 | WIMPy | ~freepbx |
| 15:24.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:24.54 | sarelon | tyvm |
| 15:25.01 | _omer | <irroot> : thanks |
| 15:27.08 | *** part/#asterisk sarelon (~sarelon@24-116-133-114.cpe.cableone.net) |
| 15:28.31 | WIMPy | Guizmo: Not too bad. But I don't know of a way to check if the line is active and the issue seems to still exist. |
| 15:28.58 | Guizmo | WIMPy: Iis there a way to force the activation ? |
| 15:29.08 | Guizmo | just to see if it is that ? |
| 15:29.19 | WIMPy | Guizmo: You can try 'pri set debug 1 span 1' (if it is our first span) and see if anything happens. |
| 15:29.50 | WIMPy | Guizmo: I don't know any with dahdi. |
| 15:30.14 | WIMPy | It sould clear when you receive a call from the net. |
| 15:31.25 | Guizmo | even locally ? I don't have a sip account on the net, and the main firewall will block it |
| 15:31.43 | WIMPy | The PSTN(et) |
| 15:35.15 | Guizmo | the debug only show in loop: http://pastebin.com/Dn1wMQTg , even when I call from PSTN. Should I try with mISDN instead of dahdi ? I have a NT box between the isdn line and asterisk. can it be the problem ? |
| 15:36.26 | WIMPy | It does say TEI assigned, so it must be talking to the switch. |
| 15:36.38 | WIMPy | Unless it's lying. |
| 15:37.24 | WIMPy | I prefer misdn2/LCR but it depends on your situation which solution is best. |
| 15:37.37 | Guizmo | will check what is the MDL-ERROR |
| 15:38.14 | WIMPy | You could try debug 2 instead of debug 1. |
| 15:49.23 | ruyo | Guizmo, do you have the jumper on TE? :P |
| 15:52.07 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:52.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:53.20 | Guizmo | ruyo: the jumper is on TE yes |
| 15:53.24 | cj | moo |
| 15:53.38 | ruyo | Just checking, you never know. :> |
| 15:53.45 | WIMPy | Early media seems broken again as well :-( |
| 15:55.57 | cj | carrar: I picked up a copy of ARRL's VoIP booklet (978-0-87259-143-1) |
| 15:55.58 | cj | http://www.arrl.org/shop/VoIP-Internet-Linking-for-Radio-Amateurs/ |
| 15:56.07 | cj | it's cute, but not very descriptive |
| 15:56.35 | cj | it covers app_rpt in 7 pages |
| 15:58.11 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
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| 16:02.19 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:02.20 | [sr] | howdy |
| 16:02.31 | [sr] | asterisk time uses the OS system time? |
| 16:04.56 | *** join/#asterisk The_REV (~The_REV@76-217-184-220.lightspeed.frsnca.sbcglobal.net) |
| 16:14.28 | WIMPy | yes |
| 16:14.45 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 16:16.52 | *** join/#asterisk nightrid3r (~kvirc@91.176.167.233) |
| 16:21.24 | [sr] | hi WIMPy |
| 16:21.25 | [sr] | hum ok.. |
| 16:21.33 | *** join/#asterisk reber (~reber@212-198-54-60.rev.numericable.fr) |
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| 16:48.01 | [sr] | brb |
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| 18:34.22 | Gugge | my asterisk sometimes just hangs. it doesnt answer sip packets, and a core stop now hangs the console. How do i see what is going on when it happens? |
| 18:35.09 | russellb | Gugge: sounds like a deadlock. First, make sure you are running the latest version. |
| 18:35.23 | russellb | If so, recompile with DEBUG_THREADS enabled (compile time option in menuselect) |
| 18:35.29 | russellb | then when it locks up, run *CLI> core show locks |
| 18:35.32 | russellb | and submit that in a bug report |
| 18:35.41 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:36.49 | Gugge | super |
| 18:36.59 | ChannelZ | or make sure your disk ain't full :) |
| 18:37.03 | Gugge | im running the latest 1.6.2 on this particular machine |
| 18:37.12 | Gugge | theres a lot of free space :) |
| 18:37.22 | russellb | ok, well then don't submit a bug report ... we're not supporting 1.6.2 anymore |
| 18:37.26 | russellb | sorry |
| 18:37.37 | Qwell | isn't sorry |
| 18:37.44 | russellb | troll! |
| 18:41.06 | *** join/#asterisk scalex000 (~chatzilla@186.6.27.121) |
| 18:41.38 | Gugge | i guess i should upgrade, but i dont really have the time :P |
| 18:41.43 | scalex000 | hello, guys, this is possible exten=>_1[N1-79X]XXXXXXXX |
| 18:41.54 | russellb | Gugge: well I wouldn't recommend it ... if it was working |
| 18:42.03 | Qwell | scalex000: I don't understand what you think that would do... |
| 18:42.06 | scalex000 | no |
| 18:42.08 | scalex000 | i get it |
| 18:42.17 | scalex000 | I found my mistake |
| 18:42.33 | russellb | the N and X in the brackets probably don't mean what you think |
| 18:42.44 | scalex000 | thank you\ |
| 18:43.00 | scalex000 | :( |
| 18:45.13 | Qwell | scalex000: also, it looks like you're trying to block 8XX numbers, or something? |
| 18:45.55 | scalex000 | hold on, I need to block 1 number 8 |
| 18:46.16 | scalex000 | http://pastebin.com/2C0Fm18z |
| 18:46.42 | scalex000 | but when I dial this number 16463186146 |
| 18:46.45 | Qwell | What are you wanting? |
| 18:46.50 | scalex000 | extension not found |
| 18:47.32 | scalex000 | I believe this pattern work |
| 18:47.43 | scalex000 | I follow some examples on voip.or |
| 18:47.48 | Qwell | I assume you're trying to send everything but tollfree numbers to Skype? |
| 18:50.28 | scalex000 | I have subcription |
| 18:50.32 | scalex000 | but my question is |
| 18:50.39 | scalex000 | why not found this extension |
| 18:50.45 | scalex000 | I think need to match |
| 18:50.48 | Qwell | because it doesn't match that pattern |
| 18:50.51 | Qwell | What are you trying to do? |
| 18:50.56 | Qwell | I've asked like 4 times now.. |
| 18:54.37 | Qwell | huh. does NANP forbid a 9 as the second digit of an area code or something? |
| 18:55.05 | russellb | I think think so ... |
| 18:55.19 | russellb | I've always seen NXXNXXXXXX |
| 18:55.38 | russellb | s/think/don't/ |
| 18:55.47 | Qwell | yeah, but I think you could use N[0-8]XNXXXXXX instead |
| 18:55.57 | Qwell | http://en.wikipedia.org/wiki/List_of_NANP_area_codes |
| 18:56.09 | Qwell | None of those actually like N9X |
| 18:56.12 | Qwell | list* |
| 18:56.14 | scalex000 | I want to call only united states area |
| 18:56.28 | Qwell | scalex000: _1NXXNXXXXXX |
| 18:56.30 | Qwell | done and done |
| 18:56.31 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:56.39 | scalex000 | but my country is 809 |
| 18:56.42 | scalex000 | 829 |
| 18:56.44 | scalex000 | 849 |
| 18:56.54 | Qwell | That's great. Your country code isn't 1. |
| 18:57.06 | scalex000 | but mobile phone does |
| 18:57.10 | irroot | prefers to use REGEX function and look up the pattern from a database :P |
| 18:57.44 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 19:10.13 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.33) |
| 19:11.22 | SpiderMon | I am having some dtmf issues Asterisk 1.6.2.13 Sangoma A400 - Outbound calls are only going through about 50% of the time |
| 19:12.05 | SpiderMon | if you call a banks ivr .. it only recognises dtmf pressed about 50% of the time |
| 19:12.27 | SpiderMon | Phones are all Polycom IP320 firmware 3.3.1 |
| 19:12.50 | SpiderMon | dtmfmode=rfc |
| 19:13.28 | SpiderMon | dtmfmode on extensions are rfc2833 |
| 19:13.55 | SpiderMon | I have dtmfmode set to info on chan_dahdi or else it gets worse |
| 19:14.06 | SpiderMon | ne suggestions? |
| 19:15.38 | irroot | try relaxed dtmf analogue channels that have not been configured properly can cause problems check impedance settings if not in USA |
| 19:15.58 | SpiderMon | in USA - relaxdtmf did not make a difference |
| 19:16.19 | SpiderMon | also toneduration=300 no go |
| 19:16.23 | irroot | check line levels if its too high or soft it can have problems |
| 19:16.39 | SpiderMon | rxgain and txgain both at 0 |
| 19:17.09 | irroot | use dahdi_monitor to look at a voice call |
| 19:17.21 | irroot | if there is echo it causes chaos with DTMF |
| 19:19.25 | SpiderMon | dahdi_monitor Rx 95-99 Tx 0 and the line is on hook |
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| 19:30.58 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.33) |
| 19:31.09 | SpiderMon | sorry my irc client keeps crashing... |
| 19:32.04 | SpiderMon | as I was say rx-95-99 on hook tx-0.. in a call rx-670-700 tx-6712 |
| 19:38.47 | SpiderMon | think i should lower the rx maybe? |
| 19:38.57 | SpiderMon | into the -5 area? |
| 19:39.05 | SpiderMon | sorry the tx |
| 19:40.59 | *** join/#asterisk scalex000 (~chatzilla@186.6.170.128) |
| 19:41.05 | scalex000 | hello, |
| 19:41.11 | scalex000 | my internet dc |
| 19:41.38 | scalex000 | someone told me to use REGEX func to check pattern |
| 19:51.57 | scalex000 | hello |
| 19:57.48 | *** join/#asterisk Dovid (4257076a@gateway/web/freenode/ip.66.87.7.106) |
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| 20:19.01 | SpiderMon | ne help on dtmf issues? |
| 20:22.04 | linuxgecko | anyone familiar with rtp300's to help me diagnose what i tink is a simple picnic error? |
| 20:30.26 | *** join/#asterisk voxter (~hardcore@macpro.daytonhome.voxter.net) |
| 20:30.39 | voxter | is the only option for having music on hold not start from the beginning of the file each time to use mpg321? |
| 20:32.03 | p3nguin | The files will be played from beginning to end. If you put someone on hold after the song has already been started, you'll hear it from the point it is rather than from the beginning. |
| 20:32.13 | voxter | nod |
| 20:32.27 | p3nguin | You're wanting to hear it from the beginning every time? |
| 20:32.56 | voxter | :Q! |
| 20:32.57 | voxter | oops. |
| 20:33.31 | voxter | p3nguin: no, I was looking for the "join the current spot in system-wide MoH" |
| 20:33.37 | voxter | ala mpg123's method |
| 20:33.45 | voxter | but, mpg123 cannot be the best way to accomplish that. |
| 20:33.47 | p3nguin | mpg123 already does that. |
| 20:34.01 | voxter | yeah, but it also has problems with consuming cpu and such |
| 20:34.13 | voxter | I mean, It works, I was just curious if there was a new "accepted" method for it |
| 20:34.45 | p3nguin | Then you need a better CPU. I ran an 800 MHz PBX (with Asterisk) and I never see any problem with mpg123 running 24 hours a day. |
| 20:35.03 | p3nguin | currently run... not "ran." |
| 20:35.37 | p3nguin | Perhaps you are transcoding a lot of calls? |
| 20:35.56 | voxter | p3nguin: I had experienced cases before not with cpu utilization of its task, but with runaway 100% cpu mpg123 processes. It was probably just a bad build. few years ago. |
| 20:37.04 | p3nguin | I'm using mpg123 1.13.2 now. |
| 20:37.25 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 20:38.11 | pzn | how to disable the "two calls" on a sip client? I mean the beep that sounds during a call telling that there is another call waiting |
| 20:38.24 | p3nguin | I think moh wav files might start and stop with calls accessing moh. That may be a way around your problem. But I'd rather fix the mpg123 problem with something not broken. |
| 20:39.14 | p3nguin | pzn: I use call-limit in the peer entry, but I think the new preferred method is group count. |
| 20:49.00 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 20:49.39 | banditti | I need to find sccp firmware for a Cisco 7960/7914. Any ideas? |
| 20:49.55 | p3nguin | Cisco has those. |
| 20:50.15 | banditti | OK, for a guy that doesn't have a cisco contract? |
| 20:50.45 | p3nguin | Just a moment. |
| 20:54.47 | p3nguin | banditti: Google for the file name. Use this as the search term for the 7960: cmterm-7940-7960-sccp |
| 20:54.56 | p3nguin | I'll see if I can find out the name for the 7914. |
| 20:55.47 | p3nguin | banditti: For the 7914, use: cmterm-7914-sccp |
| 20:56.15 | p3nguin | If you are unable to find them on your own, you'll need to pay fore the smartnet contract. :/ |
| 20:57.35 | banditti | Thanks, that helps alot! |
| 20:57.44 | p3nguin | I thought it might. |
| 21:01.49 | linuxgecko | ekiga and sipdroid can both connect to my asterisk with my test channel. but my rtp300 can't. i can only assume i have the rtp300 configured wrong. but i have yet to find instructions on how to configure it right. |
| 21:05.22 | banditti | is sccp pretty reliable on asterisk? |
| 21:08.12 | p3nguin | banditti: That depends. |
| 21:08.22 | banditti | ;) |
| 21:08.24 | banditti | On? |
| 21:08.31 | p3nguin | banditti: chan_skinny (that comes with Asterisk) SUCKS. |
| 21:08.49 | p3nguin | banditti: chan_sccp-b is pretty okay, but not available for Asterisk 1.8 branch yet. |
| 21:09.09 | p3nguin | I use chan_sccp-b on Asterisk 1.4 branch because it works. |
| 21:09.16 | banditti | is there a better sip option for a sidecar / receptionist solution? |
| 21:09.33 | p3nguin | Polycom phones, probably. |
| 21:10.36 | pzn | p3nguin, nice! call-limit worked like expected! |
| 21:11.39 | p3nguin | pzn: If you're using 1.8 branch, look into group count. People will be less likely to tell you that you're doing it wrong if you use group count instead. |
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| 22:41.47 | hetii | Hello :) |
| 22:53.58 | *** part/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:13.42 | *** part/#asterisk hetii (~hetii@87.99.51.172) |
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| 23:53.19 | Beltechs | hello im using asterisk 1.8 and am using a spa8000 at a remote location using pfsense fw at both ends running ipsec vpn. My outbound calls drop after 10 seconds. Any ideas? Thank You. |
| 23:55.19 | nightrid3r | sounds like a vpn issue |
| 23:55.37 | nightrid3r | but then i'm not an expert on vpn |
| 23:58.29 | Beltechs | so at some point the VPN is killing the call?! and causing asterisk to send a goodbye |
| 23:58.31 | Beltechs | ? |