| 00:08.43 | *** join/#asterisk drachenfels (~king@91.142.34.54) |
| 00:10.26 | drachenfels | I'm having the weirdest one way audio issue. Can anybody help? I've run wireshark and have ruled out nat and firewall. The issue only appears if I don't do a Answer on a call. |
| 00:10.58 | drachenfels | usually, on an incoming call, I just dial the extension it should go to |
| 00:11.28 | drachenfels | if they pick up the call gets connected, so there's no need for an explicit answer |
| 00:12.04 | drachenfels | The call traces with an without the answer look pretty much identical too |
| 00:13.14 | drachenfels | However, I can see an RTP packet with a mark set. but I don't really understand what this is supposed to mean. Does anyone know? |
| 00:13.38 | drachenfels | On the call without the explicit answer, I see a single RTP packet with a mark |
| 00:14.03 | drachenfels | On the call with the answer, I see two RTP packets with a mark - that's about the only difference I've been able to spot |
| 00:14.58 | drachenfels | I'm using an older release of asterisk - v 1.6.2.24, I've not really had any problems with it apart from this. |
| 00:17.00 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 00:26.07 | igcewieling | define "mark" |
| 00:27.59 | drachenfels | If you look at a packettrace, and expand out the RTSP, within there you get Stream Setup by SDP - with a bit flag field. et. but I don't really understand what this is supposed to mean. Does anyone know? |
| 00:27.59 | drachenfels | <drachenfels> On the call without the explicit answer, I see a single RTP packet with a mark |
| 00:27.59 | drachenfels | <drachenfels> On the call with the answer, I see two RTP packets with a mark - that's about the only difference I've been able to spot |
| 00:27.59 | drachenfels | <drachenfels> I'm using an older release of asterisk - v 1.6.2.24, I've not really had any problems with i |
| 00:28.11 | drachenfels | sorry.... |
| 00:28.20 | drachenfels | read the first line only |
| 00:28.54 | drachenfels | Within the bit flag field - I've got 10.. .... = Version: RFC 1889 Version (2) |
| 00:29.09 | drachenfels | ..0. .... - Padding: False |
| 00:29.22 | drachenfels | ...0 .... = Extension: False |
| 00:29.42 | drachenfels | .... 0000 = Contributing source identifiers count: 0 |
| 00:29.52 | drachenfels | 1... .... = Marker: True |
| 00:30.09 | drachenfels | it's the last line - Marker: True that I don't understand |
| 00:30.23 | drachenfels | and that I think is the cause of the problem |
| 00:30.30 | drachenfels | possibly .... |
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| 00:34.52 | igcewieling | I can't really help with rtp, but I assume you are NOT using bindport or bindaddr, correct? |
| 00:35.03 | drachenfels | RFC 1889 in section 5.3 says: The marker bit and payload type field carry profile-specific |
| 00:35.03 | drachenfels | <PROTECTED> |
| 00:35.03 | drachenfels | <PROTECTED> |
| 00:35.03 | drachenfels | <PROTECTED> |
| 00:35.03 | drachenfels | <PROTECTED> |
| 00:35.03 | drachenfels | <PROTECTED> |
| 00:35.04 | drachenfels | <PROTECTED> |
| 00:35.04 | drachenfels | <PROTECTED> |
| 00:35.04 | drachenfels | <PROTECTED> |
| 00:35.05 | drachenfels | <PROTECTED> |
| 00:35.21 | igcewieling | don't flood the channel, use pastebin |
| 00:35.30 | drachenfels | k, sorry, |
| 00:35.53 | igcewieling | have you tried directmedia=no ? |
| 00:37.13 | igcewieling | a major difference between asterisk answering and the extension answering is that if asterisk answers before sending the call to the device then media goes through Asterisk |
| 00:38.24 | *** join/#asterisk edve (~edve@modemcable062.4-22-96.mc.videotron.ca) |
| 00:40.36 | drachenfels | I just double checked, I had canreinvite=no set. Have changed it to directmedia=no but still the same issue. |
| 00:40.52 | drachenfels | The packet traces I've done appear to show everything is fine. |
| 00:41.08 | drachenfels | It only happens on certain inbound calls. |
| 00:41.54 | drachenfels | I've been working with our DID provider on this, they've said that everything looks normal to them as well. |
| 00:42.36 | drachenfels | The call goes from Certain mobile network -> DID provider -> Us |
| 00:42.56 | drachenfels | The mobile callers can't hear the voip user on our network. |
| 00:43.54 | drachenfels | They can if we do an answer. So, something on our side must be signalling our DID provider, and I think the marker is it. |
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| 00:44.43 | drachenfels | The packet traces show we are sending and receiving both RTP streams |
| 00:45.24 | drachenfels | Our provider did say that they have some new cisco gateways. |
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| 00:54.32 | *** join/#asterisk psilikon (~psilikon@173.169.199.70) |
| 00:57.28 | psilikon | Anybody in the UK right now? |
| 00:58.49 | drachenfels | yes |
| 01:05.37 | psilikon | here is the thin, my boss wanted me to grab a toll free number so our customers in the UK don't have to pay tolls. I got a 'freephone' number from a ITSP called inphonex. The number came with the exit code and country code (01144) and my boss is now worried. |
| 01:05.42 | psilikon | thin=thing |
| 01:06.48 | drachenfels | that's an area code for sheffield |
| 01:06.52 | drachenfels | not toll free. |
| 01:07.03 | psilikon | How can I be sure people from the UK won't be charged |
| 01:07.20 | drachenfels | toll free are 0800 numbers |
| 01:07.27 | psilikon | Single TOLL FREE UK DID is what is on the invoice. |
| 01:07.44 | psilikon | Yes it is an 800 number. |
| 01:08.20 | drachenfels | sorry, I'm confused by what you mean abou 01144 then |
| 01:08.26 | psilikon | Just, when they invoice my company they include the United States exit code of 011 followed by the UK country code of 44, then the 800. |
| 01:09.17 | drachenfels | I'd just email them why this is then. |
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| 01:10.30 | psilikon | The number is "8000XXXXXX". Would that be toll free for you to call? |
| 01:11.27 | drachenfels | out of interest, how much are they charging you for that? and yes in the uk, that would be free for me to call inside the UK, some mobile providers charge for 0800's but from a landline they are free. |
| 01:12.08 | psilikon | They are talking about $0.08/minute USD |
| 01:12.32 | drachenfels | that's a bit on the pricey side... |
| 01:13.08 | psilikon | drachenfels: Is there an ITSP that I can link up with that can give me a toll free number for our UK customers at a cheaper rate? |
| 01:14.00 | drachenfels | I would say us, but our billing platform is not ready yet. I'm in the middle of writing it at the moment :) |
| 01:15.06 | psilikon | drachenfels: We need to talk. We are starting to do all types of business in the UK. That's what prompted all of this. My boss called me freaking out a little while ago and trying to convince me that I had it all wrong. |
| 01:15.14 | igcewieling | I'm pretty sure we can provide a UK toll free, but it would just be resold from some UK provider |
| 01:15.29 | drachenfels | We're are a uk provider... |
| 01:15.31 | drachenfels | :) |
| 01:16.09 | igcewieling | psilikon: Likely it is a USA provider and their billing system can't handle it, so the stick a 01144 on the front of it for billing |
| 01:16.31 | igcewieling | I know our billing system gets upset when we bill for non-usa numbers. |
| 01:16.55 | psilikon | igcewieling: interesting. Excellent point. Thank you for that. |
| 01:18.39 | igcewieling | drachenfels: how many TNs are on your platform? |
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| 01:29.58 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:30.23 | saint_ | back |
| 01:30.26 | saint_ | do any of you own a business where you sell / install phone systems running asterisk ..? |
| 01:31.09 | drachenfels | Yes saint_, we do a small amount |
| 01:31.29 | saint_ | drachenfels: i m interested into getting in it on the side.. |
| 01:31.29 | igcewieling | yes. |
| 01:31.30 | drachenfels | sorry, ice - TNs? |
| 01:31.37 | igcewieling | drachenfels: telephone numbers |
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| 01:31.50 | drachenfels | ah... ok, around 10,000 |
| 01:32.05 | drachenfels | give or take |
| 01:32.11 | saint_ | igcewieling / drachenfels so you do it all over sip carriers, or you have some voice T1's cards ? How much do you guys charge for maintenance ? |
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| 01:32.32 | igcewieling | *nod* We have about 5k TNs on SIP, at least that many on TDM |
| 01:32.47 | saint_ | igcewieling: what does TN stands for ? |
| 01:32.58 | igcewieling | We use Verizon and Level 3 for wholesale SIP, Verizon for wholesale TDM |
| 01:33.01 | igcewieling | saint_: telephone numbers |
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| 01:33.41 | rhce7320 | has anybody been suscessful in getting Avaya 16xx phones to regiser as SIP clients? |
| 01:33.45 | igcewieling | Handoff to the customer is usually analog to their PBX, some PRI to their PBX, and of course SIP if the PBX supports it. |
| 01:33.49 | drachenfels | atm, we're all sip - looking at ss7 interconnects atm, as we've got a potential customer that's going to mean a !lot! of traffic |
| 01:33.50 | saint_ | igcewieling: when you say "we have" , do you host Asterisk, and have your client using VOIP over the internet to your box ? Or you mean you have a park installed outside of about 5k TN ? |
| 01:34.36 | igcewieling | saint_: We strongly discourage connecting over the internet. We sell the IP service with the voice and the IP service has QoS to our main switches |
| 01:35.25 | igcewieling | we don't "sell voip", we sell telephone service, it might be delivered over a QoS'd IP circuit or via POTS or via PRI |
| 01:35.31 | saint_ | igcewieling: do you mind telling me how you calculate the maintenance for the customers ? |
| 01:35.44 | igcewieling | saint_: I have no idea. That is sales and sales is icky. |
| 01:35.45 | rhce7320 | I'm seeing the phone send a udp/1719 packet but asterisk host returns port unreachable |
| 01:36.43 | igcewieling | I HATE bring your own broadband customers (BYOB), no QoS for example. |
| 01:37.14 | rhce7320 | udp/1719 == h323gatestat |
| 01:37.52 | igcewieling | They complain when voice quality is crap and then blame us. If they had gotten IP service from us the issue would not exist. |
| 01:39.04 | igcewieling | drachenfels: our TDM service is simply resold Verizon service under CLEC contracts. It isn't all that profitable. |
| 01:39.55 | igcewieling | rhce7320: very very very few people use H323 with Asterisk. |
| 01:41.45 | rhce7320 | um, OK... I thought I found how to switch the handset to SIP (MUTE 744 #). Maybe it lost the setting. I'll check. |
| 01:42.12 | igcewieling | rhce7320: port 5060/udp is SIP signaling and registration |
| 01:42.29 | rhce7320 | igcewieling: tks |
| 01:49.33 | drachenfels | we tend to do voip over internet... we tend to switch customers over to our own ADSL lines at the same time, so we can have some control of the quality. |
| 01:50.56 | drachenfels | more often than not, customers are on-net with us, but we do have some off-net customers, as long as they have a decent connection with a decent router, quality isn't an issue |
| 01:51.07 | rue_house | are there any asterisk developers on the mailing list? |
| 01:51.18 | drachenfels | for customers with more than 4 lines, we specifically say they must have a dedicated line. |
| 01:53.30 | igcewieling | assuming they are "on-net" we have QoS from the CPE to the wholesaler |
| 01:53.55 | igcewieling | make life easier. 8-) |
| 01:56.48 | drachenfels | Has anyone had any experience with ISDN over Ethernet? |
| 01:57.04 | drachenfels | i.e. devices like this: http://www.patapsco.co.uk/Products/ISDN/ISDN_over_IP_Product_Summary.html |
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| 02:00.24 | *** part/#asterisk ircNewbz (~ircNewbz@unaffiliated/ircnewbz) |
| 02:11.16 | saint_ | is there any reason to use Asterisk Database (AstDB) instead of MySQL ? |
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| 02:19.40 | fling | is it possible to use tcp wrappers with asterisk? |
| 02:21.20 | igcewieling | saint_: astdb Just Works. |
| 02:22.08 | fling | saint_: mysql has a lot of disadvantages, google for it :P |
| 02:22.16 | fling | igcewieling: hey |
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| 02:48.55 | din3sh | gd mrning all |
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| 02:49.49 | Sean-Der | Does anyone know a good SIP client for FreeBSD? I am trying to replace Skype, I use my Asterisk server for voice calls but never video |
| 02:50.00 | Sean-Der | I used to really like Blink, but would like something already in ports |
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| 02:58.23 | fling | din3sh: hey |
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| 03:27.19 | fling | which iax2 codec has the best quality? |
| 03:28.34 | fling | sln44 is not avaliable for iax2, right? |
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| 03:43.16 | din3sh | hey fling |
| 03:43.16 | din3sh | not sure about the best quality |
| 03:43.16 | din3sh | g729? |
| 03:45.16 | jpsharp | Best quality is going to be ULAW/ALAW. |
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| 03:50.30 | fling | jpsharp: really? but it has only 8kHz :\ |
| 03:55.56 | jpsharp | Well, actually, now that I think about it, there should be G722. |
| 03:59.11 | fling | jpsharp: thanks |
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| 06:00.04 | din3sh | zzzzzZZzzzZZ |
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| 06:11.40 | ChannelZ | snores |
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| 09:08.49 | bulkorok | hi... is there any good competitor to voxbone!? |
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| 10:39.54 | chris-NB | hi |
| 10:40.32 | chris-NB | anyone knows if trustrpid=yes takes the p-asserted-identity header and use it as callerid(num) in asterisk 10.x |
| 10:40.51 | chris-NB | it seems too me like it is done that way |
| 10:41.17 | chris-NB | put trustrip=yes should only take the remote-party-id in callerid(num) or has this been changed? |
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| 10:56.35 | wdoekes | <PROTECTED> |
| 10:56.40 | wdoekes | <PROTECTED> |
| 10:57.14 | wdoekes | i.e. it's just true or false, not pai/rpid/etc.. |
| 10:58.20 | wdoekes | static int get_rpid(struct sip_pvt *p, struct sip_request *oreq) |
| 10:58.21 | wdoekes | ... |
| 10:58.27 | wdoekes | <PROTECTED> |
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| 11:11.17 | chris-NB | wdoekes, I've seen that. above the return get_pai(p, req); is a if zero from get_rpid |
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| 11:11.35 | chris-NB | so it seems, asterisk is takin pai if rpid is empty? |
| 11:12.19 | chris-NB | compared to (I know it's old) http://www.voip-info.org/wiki/view/Asterisk+SIP+trustrpid : "P-Asserted-Identity |
| 11:12.19 | chris-NB | Asterisk does nothing when it receives a P-Asserted-Identity header. It ignores it totally no matter what settings you use for "trustrpid" or "sendrpid". It does not copy it from an inbound call leg to an outbound call leg for a bridged SIP-to-SIP call." |
| 11:12.28 | chris-NB | am I right? |
| 11:14.48 | wdoekes | ~voipinfo |
| 11:14.53 | wdoekes | ~voip-info |
| 11:14.54 | infobot | i heard voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 11:15.12 | wdoekes | hrm.. I'd expect a remark about it being "out of date" most of the time |
| 11:15.21 | chris-NB | :) |
| 11:15.24 | chris-NB | jep! |
| 11:15.29 | chris-NB | think so too |
| 11:16.22 | chris-NB | but haven't found an other usefull source :/ |
| 11:16.26 | wdoekes | but yes... if trustrpid=no => do nothing, if RPID => RPID, if PAI => PAI |
| 11:16.28 | chris-NB | expect the source |
| 11:16.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:16.44 | chris-NB | okay |
| 11:17.11 | chris-NB | if trustrpid=yes, use RPID, if RPID is absent, check/use PAI |
| 11:17.12 | chris-NB | right? |
| 11:28.40 | wdoekes | that's the ordering indeed |
| 11:29.14 | wdoekes | although you may need some parentheses around your sentence: |
| 11:29.27 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 11:29.44 | wdoekes | if trustrpid=yes (use RPID (if RPID is absent, check/use PAI)) |
| 11:30.57 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 11:32.05 | *** join/#asterisk dkostousov (~dkostouso@78.159.236.154) |
| 11:32.28 | dkostousov | Hi all. Does asterisk play file from http url? |
| 11:41.43 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 11:44.31 | *** join/#asterisk mrkon (~mrkon@176.57.72.74) |
| 11:44.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:49.45 | wdoekes | dkostousov: you'd have to download it first (see func_curl) |
| 12:00.54 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 12:06.31 | *** join/#asterisk nacho2k (~Thunderbi@r190-64-14-98.ir-static.anteldata.net.uy) |
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| 12:27.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 12:54.11 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-136-163.user.veloxzone.com.br) |
| 12:56.28 | Rico29 | hi all |
| 12:56.32 | Rico29 | is there anybody here ? |
| 12:56.37 | anonymouz666 | nops |
| 12:59.23 | *** join/#asterisk timahvo1 (~rogue@197.177.55.243) |
| 12:59.24 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 12:59.48 | Rico29 | I have a strange problem |
| 13:00.14 | Rico29 | I have a file with this permissions : -rwxrwx--- 1 apache asterisk_sounds 314868 déc. 5 16:52 asterisk_sounds/closed/oceanet/Accueil2011.wav |
| 13:00.39 | Rico29 | and users asterisk and apache are members of group "asterisk_sounds" |
| 13:00.53 | Rico29 | but asterisk process can not read the file |
| 13:01.24 | Rico29 | if I manually login with user asterisk (su - asterisk), I can read the file |
| 13:01.44 | Rico29 | if I change owner of file (s/apache/asterisk/), it works |
| 13:02.19 | Rico29 | as if my asterisk process can not access files within it's group.. |
| 13:02.40 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.204) |
| 13:03.04 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:10.18 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:12.00 | Rico29 | any idea ? |
| 13:12.23 | plundra | At what latencies will SIP start to get annoying (requiring tweaks and "help")? Currently the RTT is ~10ms avg. (+/- 3ms) between the office in question and the asterisk-box in a datacenter; would going up the double, 20ms, matter? |
| 13:12.52 | Rico29 | no |
| 13:12.58 | kaldemar | plundra: 20 ms is really nothing at all. |
| 13:13.01 | Rico29 | but depends on jitter to |
| 13:13.10 | bulkorok | we have working SIP/RTP with >200ms |
| 13:13.12 | Rico29 | 3ms is nothing |
| 13:14.07 | Rico29 | any idea for my permissions problem ? |
| 13:14.10 | *** join/#asterisk timahvo1 (~rogue@197.177.55.243) |
| 13:17.42 | kaldemar | Rico29: is asterisk running as a user called "asterisk"? |
| 13:18.32 | kaldemar | Rico29: is the path readable for asterisk as well? |
| 13:20.11 | *** join/#asterisk evilman_home (kvirc@95-29-217-32.broadband.corbina.ru) |
| 13:21.14 | Rico29 | kaldemar> yes, asterisk runs in user asterisk / group asterisk |
| 13:21.23 | Rico29 | and full path is readable by group |
| 13:24.14 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:26.02 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 13:28.15 | kaldemar | Rico29: "asterisk runs in user asterisk / group asterisk" <-- group mismatch maybe, the group should be asterisk_sounds. |
| 13:31.38 | Rico29 | kaldemar> I don't think so |
| 13:32.24 | Rico29 | asterisk process tries to access a file that is owned by group "asterisk_sounds", and user "asterisk" belongs to group "asterisk_sounds" |
| 13:33.15 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 13:33.22 | wasanzy | hi |
| 13:33.43 | wasanzy | am facing trouble with some modules |
| 13:34.10 | plundra | Rico29, kaldemar: Okay great. Yeah no the 20ms-one is rock solid at too, more so than the other, currently at least. Jitter-wise. Thanks! |
| 13:34.40 | kaldemar | Rico29: what does asterisk say in the first place when it tries to read it? and you're welcome to make a pastebin that shows all the permissions and groups. |
| 13:35.02 | Rico29 | kaldemar> ok |
| 13:35.33 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:36.06 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 13:39.11 | leifmadsen | does anyone in here have access to a 32-bit installed version of Ubuntu 12.04? |
| 13:39.12 | *** join/#asterisk hbee (~hbee@87-100-155-185.bb.dnainternet.fi) |
| 13:39.18 | leifmadsen | I need something looked up for the Asterisk book |
| 13:39.25 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:39.25 | leifmadsen | (you will get credit!) |
| 13:40.51 | kaldemar | yup |
| 13:41.24 | kaldemar | what do you want to know? |
| 13:41.42 | Rico29 | kaldemar> http://pastebin.com/22dPhSE2 |
| 13:42.10 | leifmadsen | /usr/lib/x86_64-linux-gnu/ exists on 64-bit versions; on 32-bit is it /usr/lib/i386-linux-gnu? |
| 13:42.11 | Rico29 | no, sorry leifmadsen |
| 13:42.37 | leifmadsen | ideally, apt-get install tdsodbc ; find / -name "libtdsodbc.so" |
| 13:43.54 | kaldemar | leifmadsen: /usr/lib/i386-linux-gnu . /usr/lib/x86_64-linux-gnu exists (comes from the cpp package) but it is empty. |
| 13:44.10 | leifmadsen | kaldemar: sure, probably since nothing installed to those locations, thanks much |
| 13:44.17 | leifmadsen | kaldemar: real name you prefer in the book? |
| 13:45.04 | kaldemar | mind if i pm that? |
| 13:45.22 | leifmadsen | not at all |
| 13:50.44 | [TK]D-Fender | Rico29, I looked at your pastebin suspecting I'd see something off but it looks pretty thorough, except for one thing : verify exactly who Asterisk is running under. It's the one thing we don't see |
| 13:50.45 | wdoekes | apt-cache show tdsodbc | grep Multi-Arch && echo goes in `uname -m` |
| 13:50.58 | wdoekes | @ leifmadsen |
| 13:51.10 | leifmadsen | wdoekes: clever :) |
| 13:51.36 | Rico29 | [TK]D-Fender> asterisk 14605 14603 1 2012 ? 14:28:53 /usr/sbin/asterisk -f -vvvg -c |
| 13:51.45 | leifmadsen | found it on packages.ubuntu.com so I'm all set |
| 13:51.54 | anonymouz666 | if you set a a T1 port to loop through dahdi_maint and frame errors keeps increasing means exactly what? |
| 13:52.25 | anonymouz666 | dahdi_maint = logical loopback |
| 13:55.58 | *** join/#asterisk serafie (~erin@nat/digium/x-pawdtibtkboarfbm) |
| 14:03.48 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:04.23 | *** join/#asterisk rjvvliet (~rjvvliet@2001:610:600:86a5:226:b9ff:fe96:d6c2) |
| 14:11.36 | [TK]D-Fender | Rico29, Rico29 And asterisk.conf? |
| 14:12.02 | Rico29 | I'm pastebining it right now |
| 14:12.47 | Rico29 | but if the question is about runuser and rungroup, answer is "asterisk" |
| 14:12.54 | Rico29 | [TK]D-Fender> |
| 14:13.43 | [TK]D-Fender | hrmm... |
| 14:14.25 | *** join/#asterisk timahvo1 (~rogue@197.177.55.243) |
| 14:17.14 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.204) |
| 14:19.56 | kaldemar | Rico29: is rungroup also asterisk? |
| 14:20.41 | Rico29 | <PROTECTED> |
| 14:20.51 | [TK]D-Fender | kaldemar, Indeed, if you have to specify the group does that mean that it doesn't follow the perms of the user? |
| 14:21.03 | kaldemar | [TK]D-Fender: yes. |
| 14:21.17 | [TK]D-Fender | kaldemar, Can you specify multiple? |
| 14:21.26 | [TK]D-Fender | Rico29, This would seem to be the issue |
| 14:21.30 | kaldemar | it uses the defined group only, unless i'm mistaken. |
| 14:21.43 | Rico29 | so I have to remove rungroup ? |
| 14:21.45 | [TK]D-Fender | kaldemar, But can you specify multiple? rungroup=a,b,c? |
| 14:21.55 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-218-179.red.bezeqint.net) |
| 14:23.17 | kaldemar | no, it's one group only. |
| 14:24.33 | Rico29 | and if you don't specify a arungroup ? |
| 14:25.16 | kaldemar | Rico29: comment the setting out and you should be good. |
| 14:25.44 | Rico29 | kaldemar> ok, will try as soon as possible (prod env.) |
| 14:28.51 | kaldemar | looks like the parameter is for limiting the group to a single group only. |
| 14:30.39 | Rico29 | ok |
| 14:39.56 | *** join/#asterisk cheasee (~cheasee@vie-188-118-246-206.dsl.sil.at) |
| 14:41.51 | igcewieling | the operating system likely is limiting it to one group. |
| 14:43.33 | kaldemar | you'll see multiple groups with !groups in CLI but rungroup limits permissions to a single one. |
| 14:46.08 | igcewieling | interesting. |
| 14:48.13 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 14:49.05 | *** join/#asterisk timahvo1 (~rogue@197.177.55.243) |
| 14:57.13 | Kobaz | mmm |
| 14:57.13 | Kobaz | run |
| 14:57.15 | Kobaz | fun |
| 14:57.28 | Kobaz | reloading 400 sip peers makes all the peers go unreachable for 1-2 seconds |
| 14:57.46 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 14:58.29 | nunne | Does anyone know if \ works as character escape in sip.conf? Need to have ";" in my secret. So would "\;" work? |
| 14:59.02 | Kobaz | sounds like a bad character to have as a secret |
| 14:59.05 | igcewieling | Kobaz: what version of Asterisk |
| 14:59.13 | Kobaz | igcewieling: 1.8.12 |
| 14:59.39 | igcewieling | interesting, that is something I might expect with 1.4, but not 1.8. |
| 15:00.26 | MarkS- | nice task for my weekend, find a way to use the agi option for asterisk (or is there another way to let php make some decisions regarding call routing?) |
| 15:00.33 | Kobaz | i need to start playing with 11 one of these days |
| 15:00.45 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:00.47 | Kobaz | this is the first major issue i've run into with 1.8 in a while though |
| 15:01.20 | nunne | Never mind, \ worked as an escape char :) |
| 15:01.36 | nunne | Kobaz: indeed, but sometimes providers will give you and password.. and thats it :P |
| 15:01.37 | Kobaz | that's good to know |
| 15:01.59 | Kobaz | you can usually get them to generate something new |
| 15:04.02 | [TK]D-Fender | MarkS-, "core show application SHELL" |
| 15:04.05 | kaldemar | MarkS-: func SHELL and app System |
| 15:04.07 | nunne | Kobaz: after being stuck on hold for like 2h my colleage gave up.. and I tried this approach instead. problem solved :D Seems like ISP personell don't like working on friday evening ;D |
| 15:04.47 | [TK]D-Fender | sorry, meant "function", not "application" |
| 15:05.19 | [TK]D-Fender | MarkS-, SHELL should allow you to get a value back from your script, but isn't "interactive, just a 1-way passing of vars in and result out |
| 15:05.26 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 15:05.28 | [TK]D-Fender | MarkS-, Aside from that, AGI is the way to go. |
| 15:06.02 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 15:06.53 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 15:08.14 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 15:08.19 | MarkS- | [TK]D-Fender: thanks for mentioning it, I will check it. I need it to make a decision: allow calling or not |
| 15:09.21 | [TK]D-Fender | MarkS-, SHELL might do it then depending on how much you need to pass for validation. |
| 15:11.39 | MarkS- | only the used sip account is required (for us) |
| 15:12.00 | MarkS- | maybe in the future the number that is called |
| 15:12.30 | [TK]D-Fender | MarkS-, I'd probably just do an AGI for this then.... will save you future conversion grief |
| 15:13.29 | igcewieling | MarkS-: In my experience doing an actual DIAL from AGI is not a good idea. What we do is use the AGI to set channel variables and handle the dialing in the actual dialplan? |
| 15:13.54 | MarkS- | I will check both options |
| 15:14.29 | nunne | MarkS-: I have done alot of call routing with AGI PHP and bash scripts.. And having the script return channel variables.. works like a charm |
| 15:15.10 | MarkS- | any examples I should check? |
| 15:15.32 | [TK]D-Fender | ~book |
| 15:15.32 | 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 |
| 15:15.34 | [TK]D-Fender | ^ |
| 15:16.03 | [TK]D-Fender | All in there. And there is the PHP-AGI library that's ready to use and makes a good amount of the calls very easy with tons of samples out there |
| 15:16.41 | igcewieling | If you are using AGI the only reason you would not use an AGI library for your chosen language is if you like pain, suffering, and frustration. |
| 15:16.50 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.204) |
| 15:17.43 | MarkS- | I will check it, with this I should have enough information to do it |
| 15:18.15 | MarkS- | I sometimes like pain ;) for sending mails with php I have an own smtp function (not using phpmailer/mail())... |
| 15:19.26 | nunne | MarkS-: not where I am right now.. at a customers place now ;) but it's rather simple |
| 15:20.12 | nunne | in bash you just echo the agi command for setting channel variable.. think it's "set variable varname "value"" |
| 15:20.21 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 15:21.40 | wasanzy | I have curl, libcurl-devel, net-snmp, net-snmp-devel all installed but res_config_curl res_curl res_snmp func_curl can not be installed from menuselect |
| 15:21.48 | wasanzy | what am I suppose to do? |
| 15:21.55 | nunne | and same thing I suppose for php ;) |
| 15:22.34 | nunne | wasanzy: did you run ./configure after you installed these packages? |
| 15:26.17 | Katty | dear friends. |
| 15:26.27 | Katty | what ssh app should i be putting on this iphone? |
| 15:26.30 | Katty | Love, Katty |
| 15:27.52 | [TK]D-Fender | Katty, Spackle. It's just like PuTTY, only taupe ;) |
| 15:29.01 | wasanzy | nunne: yes I did |
| 15:29.06 | wasanzy | but still |
| 15:29.57 | *** join/#asterisk jero (~boo@mtl.savoirfairelinux.net) |
| 15:30.43 | Katty | [TK]D-Fender: that must be for ipad only |
| 15:30.43 | jero | hi |
| 15:30.45 | Katty | [TK]D-Fender: :< |
| 15:30.52 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:30.54 | Katty | no sign of it on the app store. |
| 15:30.59 | rue_house | can anyone help me with a mgcp problem? I think mgcp.conf can only define one gateway |
| 15:31.02 | Katty | shame too. |
| 15:31.32 | wasanzy | Do you think I should run ./configure --libdir=/usr/lib64 because I guess most of the libs are installed there instead of /usr/lib |
| 15:31.47 | jero | I need to put a dahdi fxs channel on hold, how can that be done ? |
| 15:32.19 | rue_house | you want to release the calls possesion to nobody? |
| 15:32.22 | rue_house | park it |
| 15:32.28 | wasanzy | ah here is one: /usr/lib64/libcurl.so guess I have to do that |
| 15:33.44 | rue_house | jero, or do you mean the fxs side set wants to put the call on hold |
| 15:33.52 | rue_house | you can only do so much with hook flash |
| 15:34.25 | rue_house | there is an inband digits thing too, dont know what features you can tie to it |
| 15:36.23 | igcewieling | nunne: incorrect. You must read your stdin from asterisk or you can't do anything |
| 15:38.11 | Katty | test from iphone |
| 15:38.18 | Katty | fancy. |
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| 15:43.24 | chuckf | do you get a 'not actual size' message on the iPhone? |
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| 15:49.25 | wasanzy | guyz I have resolved the problem, ./configure --libdir=/usr/lib64 did the magic for me |
| 15:49.30 | Katty | chuckf: why do i get the feeling that a joke just went over my head. |
| 15:50.24 | anonymouz666 | http://oxforddictionaries.com/words/the-oec-facts-about-the-language -> vocabulary size = 100 words 50% of the content in Oxford English Corpus |
| 15:51.03 | chuckf | Katty: it was a small joke... most of the time people irc from a desktop with a decent sized window, and you were irc'ing from the iphone with a small screen... |
| 15:51.21 | chuckf | or at least I was guessing you were |
| 15:51.36 | Katty | ah. |
| 15:51.43 | Katty | yes, it was a bit small ;) |
| 15:51.58 | Katty | luckily all i needed was an ssh app tho, and not an irc app |
| 15:52.08 | anonymouz666 | samsung galaxy note II |
| 15:52.28 | Katty | the boy has a samsung galaxy s iii i think |
| 15:52.29 | Katty | it's huge |
| 15:52.51 | anonymouz666 | note II I think is 5,3 |
| 15:52.56 | anonymouz666 | S III is 4,8 |
| 15:53.10 | chuckf | the sg3 is a large device. I'm wondering if I'll be happy with it when I get one |
| 15:53.11 | *** join/#asterisk deo (~deo@112.198.79.21) |
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| 15:53.27 | chuckf | Katty: does the boy like it? |
| 15:53.32 | Katty | when it works. |
| 15:53.34 | anonymouz666 | note 2 is bizarre when using as a phone |
| 15:53.44 | Katty | he says the android OS is more buggy than the iOS one |
| 15:53.46 | coppice | the S3 was big last year, but is small this year. standards change :-) |
| 15:54.20 | anonymouz666 | S3 is nice, Note 2 is better. I don't have problems with big devices |
| 15:54.23 | Katty | we found that Siri does math better than Galaxy |
| 15:54.33 | chuckf | I use android on all tablet/phone devices now so I'm used to the quirks |
| 15:54.39 | anonymouz666 | iOS is gonna be killed by android.... |
| 15:54.51 | Katty | galaxy seems to do better at navigation requests. |
| 15:55.01 | coppice | the note was a massive hit here. all my wife's friends swapped their iphones for notes |
| 15:55.44 | anonymouz666 | iOS -> android is a natural move |
| 15:55.51 | *** part/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 15:56.02 | Katty | i think it's pretty normal for someone to try out both. |
| 15:56.08 | Katty | just another toy |
| 15:56.19 | Katty | i'm getting the boy a chromebook and an android mini 4.0 |
| 15:56.30 | Katty | he will enjoy tinkering with those too |
| 15:56.32 | coppice | i think many of these people would swap back to a 5.5" iphone, but they have no interest in the curretn small ones |
| 15:56.59 | anonymouz666 | the android software is much better in my opinion |
| 15:57.02 | Katty | that big screen sure is nice for some things. |
| 15:57.12 | Katty | like remoting into a server. |
| 15:57.43 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 15:57.59 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:58.45 | anonymouz666 | Katty: i tried to use SSH client but I can't it's just too difficult |
| 15:59.04 | anonymouz666 | patience is required |
| 15:59.29 | Katty | anonymouz666: i haven't figured out how to switch channels |
| 15:59.30 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:59.35 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-218-179.red.bezeqint.net) |
| 15:59.38 | coppice | SSH is OK from a note when you really need to do something |
| 15:59.43 | Katty | the alt or ctrl n thing doesn't seem to ...function |
| 15:59.54 | anonymouz666 | with 7000 words in my vocabulary I would understand 90% of content in oxford english corpus |
| 15:59.58 | anonymouz666 | 1000 words 75% |
| 16:00.08 | anonymouz666 | I am somewhere between 1000 and 7000 |
| 16:00.14 | *** join/#asterisk Dovid (~Dovid@67.82.57.131) |
| 16:00.28 | anonymouz666 | Katty used a word out of this range :P |
| 16:00.41 | *** part/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 16:01.03 | Katty | odds are it wasn't /real/ english then hehe |
| 16:01.07 | chuckf | I've been pretty happy with the hackers kb on the android side of things for sshing |
| 16:01.44 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
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| 16:13.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:38.14 | jero | rue_house: I want to emulate a sip phone's home button on an fxs phone |
| 16:38.23 | jero | s/home/hold |
| 16:39.28 | igcewieling | jero: see features.conf |
| 16:44.57 | [TK]D-Fender | jero, There is no "hold" on an analog phone in that sense. You'll have to park the call. |
| 16:53.41 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 17:03.30 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:15.55 | *** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com) |
| 17:17.16 | mcargile | Is there a way to tell how the software timer in Dahdi is getting its timing? I am getting a timer accuracy of 4% on a completely unloaded system and wondering if there is a hardware issue. |
| 17:19.07 | Greenlight | I beleive if you've not got hardware source, it'll use the system timer |
| 17:19.13 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 17:20.02 | Greenlight | cat /sys/devices/system/clocksource/clocksource0/current_clocksource |
| 17:20.16 | Greenlight | If it's linux; I beleive that's what it'll use |
| 17:20.22 | Greenlight | COuld be wrong though |
| 17:20.26 | mcargile | tsc |
| 17:20.41 | Greenlight | cat /sys/devices/system/clocksource/clocksource0/available_clocksource |
| 17:20.45 | Greenlight | What's that give you |
| 17:21.24 | mcargile | tsc hpet acpi_pm |
| 17:21.34 | Greenlight | hpet is generally the best |
| 17:21.44 | Greenlight | What distro you using ? |
| 17:21.48 | mcargile | Ubuntu |
| 17:22.11 | Greenlight | Not sure if the kernal is 1000hz by default, as it is on CentOS - perhaps someone else can verify ? |
| 17:22.25 | mcargile | i can |
| 17:22.40 | mcargile | 100Hz |
| 17:22.50 | Greenlight | That's quite likely the problem |
| 17:23.01 | mcargile | how do I switch it to hpet? |
| 17:23.21 | Greenlight | Well, as well as switching to hpet, you need to run a 1000hz kernel |
| 17:23.35 | Greenlight | hpet is easy, add clocksource=hpet to grub |
| 17:24.24 | mcargile | my understanding was the 1000hz timing kernel was not an issue as of 4 years ago |
| 17:25.00 | Greenlight | Well that's news to me |
| 17:25.23 | mcargile | I have other Ubuntu servers runnign the same kernel without issue |
| 17:25.36 | Greenlight | Then perhaps it's a none issue |
| 17:25.43 | Greenlight | Try setting clocksource to hpet |
| 17:25.49 | mcargile | which is why I think it is a hardware issue |
| 17:25.56 | mcargile | stupid grub2 |
| 17:26.07 | Greenlight | You could be right - I'm only half sure dahdi uses the kernel timing |
| 17:26.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@91.183.118.5) |
| 17:26.29 | Greenlight | Was hoping someone would chip in and verify or correct .. |
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| 17:30.19 | igcewieling | are you sure you're not looking for ASTERISK timing? |
| 17:30.40 | igcewieling | "core show modules like tim" with an active call will show you where asterisk is getting its timing from |
| 17:30.47 | mcargile | Best: 53.507% -- Worst: 4.688% -- Average: 10.791148% |
| 17:30.54 | mcargile | from dahdi_test |
| 17:31.07 | mcargile | other servers it is 99% |
| 17:31.15 | igcewieling | mcargile: what problem are you trying to solve? |
| 17:31.17 | mcargile | switching the clocksource to hpet did not do it |
| 17:31.17 | Greenlight | Yea, that's some pretty awful timing |
| 17:31.37 | mcargile | my meetme and moh sounds like crap because my dahdi timer is crap |
| 17:31.48 | mcargile | this is asterisk 1.4 |
| 17:32.01 | igcewieling | oh! that sucks |
| 17:33.01 | Greenlight | Just something to check - are all powersaving features disabled ? |
| 17:33.35 | mcargile | Hard to check. This is a hosted server. |
| 17:34.10 | Greenlight | Is it virtualised ? |
| 17:34.38 | mcargile | no |
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| 17:36.12 | WIMPy | Even 99% is pretty bad. |
| 17:36.31 | Greenlight | Anything ynder 99.997 I'd say would cause issues |
| 17:36.32 | Greenlight | *under |
| 17:36.55 | igcewieling | why are you not using a modern Asterisk version? |
| 17:42.26 | kaldemar | mcargile: what version of DAHDI are you using? |
| 17:42.55 | mcargile | The program that were trying to use is only compatible wiht 1.4 and 1.8 support for 1.8 was added last week so I dont trust it |
| 17:42.59 | mcargile | 2.6.1 |
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| 17:44.19 | mcargile | the issue is not the asterisk version though. it is dahdi timing |
| 17:45.05 | WIMPy | Or the fact that the old Asterisk version needs dahdi timing. |
| 17:45.35 | mcargile | and if you have any problem with pthread timing the first thing anyone will tell you is to use dahdi |
| 17:45.55 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 17:45.56 | Greenlight | timerfd is quite reliable these days |
| 17:46.22 | Greenlight | Your scores are soo far out that something drastic is wrong though, imo |
| 17:46.29 | WIMPy | has never had any issues with it. |
| 17:46.43 | WIMPy | That is completely broken, yes. |
| 17:46.45 | mcargile | ahh the same dahdi should not be giving me such crap timing and I am trying to figure out if the motherboard is crap |
| 17:46.52 | mcargile | *all |
| 17:46.56 | Greenlight | I don't do much with Ubuntu so can't advise based on exp |
| 17:47.02 | mcargile | or if it compiled wrong |
| 17:47.37 | Greenlight | You're running latest available kernel ? Running same setup else where ? |
| 17:47.53 | Greenlight | My betting is something like powersaving options. That or the 100hz kernel. |
| 17:47.56 | mcargile | for this distro yes I am running the latest kernel |
| 17:48.25 | mcargile | zaptel stopped using the kernel tick for timing years ago |
| 17:48.32 | mcargile | before it became dahdi |
| 17:48.52 | Greenlight | So, how's it get it's timing ? |
| 17:49.09 | WIMPy | I'm using only 300Hz time, tickless sysyem an all power management available. |
| 17:49.24 | Greenlight | And DAHDI ? |
| 17:50.00 | kaldemar | mcargile: afaik, not having a 1000 hz kernel should not be an issue with new DAHDI versions. |
| 17:50.00 | mcargile | it is still a 2.6 kernel on this system so I am going to try and older version of dahdi |
| 17:50.02 | WIMPy | Only one box still on dahdi, because of hardware. |
| 17:50.19 | mcargile | kaldemar: agreed |
| 17:50.22 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:50.41 | mcargile | have several servers runnign this same kernel version without issue |
| 17:53.30 | igcewieling | Asterisk 1.8 (maybe 1.6) and later do not need dahdi timing, only dahdi for audio mixing |
| 17:55.06 | mcargile | igcewieling: I am aware. Asterisk 1.6 also crashes under heavy load, and Asterisk 1.8 had serious issues with MeetMe up until 1.8.19 |
| 17:56.50 | Greenlight | http://ibot.rikers.org/%23asterisk-dev/20121112.html.gz <-- someone having similarly bad scores as yourself, not had time to read it all, as gotta leave office now, but maybe worth a look |
| 17:57.05 | mcargile | thanks Greenlight |
| 17:57.09 | Greenlight | Search for "dahdi_test" |
| 17:57.41 | *** join/#asterisk cmendes0101 (~cmendes01@216.104.164.5) |
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| 18:02.38 | *** join/#asterisk tekzilla (~jon@hmbg-4d06a6d1.pool.mediaWays.net) |
| 18:03.04 | tekzilla | hey, can i get to channel vars of another SIP channel ? |
| 18:05.25 | *** join/#asterisk lifeforms (~walter@tau.lfms.nl) |
| 18:06.25 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 18:06.59 | [TK]D-Fender | tekzilla, How would youknow the other channel? When would you do this? What is the goal? |
| 18:08.37 | lifeforms | my home asterisk has a weird case of the crashes.. at some point, it stops doing iax2 and the process seems to go into a weird state.. 'core stop now' doesn't do anything, and all iax2 peers are unreachable (qualify)... I now notice from messages it always seems to happen 7 minutes after midnight http://pastebin.com/SEXRNRsG - Any debugging ideas? |
| 18:08.39 | tekzilla | i set up some CHanspy stuff today, and when capturing a recording, i would like to include the callee of the SIP channel im spying on in the filename |
| 18:09.30 | [TK]D-Fender | tekzilla, So you're pulling this externally then? Not from within another call? |
| 18:09.55 | tekzilla | well when i'm spying on someone i am on my own channel right |
| 18:10.12 | igcewieling | lifeforms: check cron for jobs running around that time. |
| 18:10.51 | tekzilla | and the sip channel name will always start with "SIP/extension" right ? |
| 18:11.13 | tekzilla | for the internal extension i'm spying on |
| 18:11.18 | igcewieling | tekzilla: no. |
| 18:11.25 | igcewieling | it will be SIP/thepeeryoudefinedinsip.conf |
| 18:11.45 | igcewieling | newbies often make the peer name the same as an extension, but that is seldom a great idea. |
| 18:11.53 | lifeforms | igcewieling: nothing really runs.. maybe newsyslog (FreeBSD log rotation) though I have not put any asterisk stuff in its config |
| 18:13.01 | [TK]D-Fender | tekzilla, I'd do this in AMI. Launch a script passing the partial match for the issuer and poll the channel the has chanspy active on it. That should get your the bridged channel |
| 18:13.41 | WIMPy | Or just have something running that listens on AMI and sees that you're invoking ChanSpy. |
| 18:16.25 | tekzilla | yeah ok i guess thats possible, thanks for the ideas |
| 18:17.17 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
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| 18:48.40 | igcewieling | lifeforms: make SURE ALL ip addresses configured in the system are listed in /etc/hosts If you internet goes down and DNS can't happen asterisk gets very up set. Maybe your internet drops at the time |
| 18:55.38 | lifeforms | igcewieling: thanks, could be a nice one :) also maybe my cable modem is resetting its NAT tables at that moment |
| 18:55.47 | lifeforms | router* |
| 18:57.09 | igcewieling | It is a common issue, though people often don't realize it because 1) they are so freaked out by their internet being down they don't focus on solving the issue or 2) anyone who knows anything if off site and can't get into the system since the internet is down. |
| 18:59.52 | lifeforms | meanwhile I've created a cron script to check it, if the issue goes away it was likely DNS :) |
| 19:00.03 | lifeforms | thanks! |
| 19:01.17 | igcewieling | no problem |
| 19:02.08 | SeRi | nice and quiet today |
| 19:02.15 | SeRi | weather is awesome |
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| 19:55.45 | macroevolve | i'm confused. i had a tech (who I can no longer access) help me setup asterisk to help take speed measurements on a call. He calculated that the "avg packet loss" was 1.2%. I have no idea how this was calculated and was wondering if anyone knows. i see in the mysql database that "received_packets" = 1145, "local_lostpackets" = 50, "remote_lostpackets" = 3, and "transmitted_packets" = 3019 |
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| 20:26.02 | Slugs_ | exit |
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| 20:51.24 | ruben231 | any chance i can connect my Avaya Phone 4620sw to my asterisk |
| 20:51.38 | ruben231 | anyone have successfully implemented this..? |
| 20:51.40 | Qwell | Does it use a protocol Asterisk supports? |
| 20:51.44 | igcewieling | ruben231: does it run standard SIP? |
| 20:51.55 | ruben231 | Qwell:im not sure with that |
| 20:52.14 | ruben231 | igcewieling:but i see H323.. |
| 20:53.08 | dpilon | SIP also |
| 20:53.14 | igcewieling | ruben231: H323 is not very common with Asterisk and you will find few people who can help with it. |
| 20:53.14 | dpilon | from the look of it |
| 20:54.02 | igcewieling | If you just want to experiement, use a softphone. If you want production get something like a Linksys SPA or Polycom |
| 20:54.12 | Qwell | Definitely get Digium phones. |
| 20:54.21 | Qwell | My totally unbiased opinion. |
| 20:55.57 | dpilon | ruben231: just make sure the firmware on the phone has sip support..if not..upgrade the firmware |
| 20:57.44 | ruben231 | dpilon: thats the problem i dont have access with the firmware to download |
| 20:58.09 | ruben231 | dpilon:can you help somehow how todo this |
| 20:58.22 | dpilon | i don't have access to that either |
| 20:58.32 | igcewieling | Qwell: are Digium phones based on any well known platform like, for example, Polycom? |
| 20:59.27 | igcewieling | We won't switch mainly because we have way way way too much invested in Polycom phones and custom provisioning for those phones. |
| 20:59.43 | Qwell | igcewieling: See; DPMA. Provisioning is not a problem whatsoever. |
| 20:59.48 | Qwell | and no they are not. |
| 21:00.08 | ruben231 | dpilon: :-( hopeless |
| 21:00.20 | Qwell | igcewieling: https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phone+Module+for+Asterisk+%28DPMA%29+v+1.4 |
| 21:00.25 | igcewieling | Qwell: provisioning is not a problem if you want to start from scratch. integrating provisioning into our custom provisioning system would be a hassle |
| 21:00.28 | navaismo | igcewieling, when you use a digium you just love it! provision its really easy |
| 21:00.42 | Qwell | igcewieling: No, you're missing the point. The provisioning is unnecessary. |
| 21:00.43 | navaismo | sine you can create templates as sip conf |
| 21:00.59 | Qwell | Write Asterisk configs, be done. |
| 21:01.08 | igcewieling | Qwell: Is the check in the mail too? I just cannot believe they "just magically work" |
| 21:01.16 | Qwell | See previous link. |
| 21:01.20 | igcewieling | Qwell: FreePBX writes the configs |
| 21:01.49 | igcewieling | our stuff reads the FreePBX database and builds config files based on that |
| 21:01.53 | Qwell | igcewieling: FreePBX writes DPMA configs too. |
| 21:02.04 | Qwell | It's automatic. There is not a caveat. |
| 21:02.28 | igcewieling | nifty. if Polycom pisses us off again I might look at it. I just hate the idea of supporting multiple vendors, even from a user standpoint |
| 21:03.21 | igcewieling | If Digium phones support key system features with Asterisk and do not crash/lag when the phone subscribes to 50 bazillion extensions, that could make us switch |
| 21:26.13 | rrittgarn | Anybody use Monitor to record all calls on a PBX, or is the general solution to use something like Oreca/OrecX? |
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| 21:57.02 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 21:57.59 | Mon|A|rch | hey, I'm trying to transfer variables out of the dialplan, and my code looks like same=>n,Set(response=${CURL("${ASERVER}/aurl?avar=avalue")}) |
| 21:58.15 | Mon|A|rch | and I'm getting the error "function CURL not registered" |
| 21:58.26 | *** part/#asterisk piffey (~pfe@gateway/tor-sasl/piffey) |
| 21:58.31 | Mon|A|rch | in menuselect, ti shows curl as being usable |
| 21:59.53 | ChannelZ | what of "core show function CURL"? |
| 21:59.58 | Qwell | Mon|A|rch: It isn't registered. |
| 22:00.43 | ChannelZ | if you get no function with that, then 'module load func_curl' and see what happens. |
| 22:00.54 | Mon|A|rch | Qwell, how do i go about registering it? I haven't really changed anything for a while, but this code worked about two weeks ago |
| 22:01.05 | Mon|A|rch | ChannelZ, alright |
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| 22:01.20 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:01.25 | Mon|A|rch | module could not be loaded |
| 22:01.49 | ChannelZ | Sounds like something failed building it the last time you recompiled perhaps. |
| 22:02.00 | Mon|A|rch | hm |
| 22:02.03 | Mon|A|rch | poop |
| 22:02.22 | ChannelZ | look in /usr/lib/asterisk/modules (typically) and see if it's even there |
| 22:02.53 | Mon|A|rch | alright |
| 22:03.38 | Mon|A|rch | ChannelZ, i see |
| 22:03.40 | Mon|A|rch | it's not there |
| 22:03.43 | Mon|A|rch | neither is app_curl |
| 22:04.26 | ChannelZ | There is no app_curl (that I know of), it's a function not a dialplan application |
| 22:04.39 | Mon|A|rch | all i really want to do is send an http request to the server, will func_url work for that? |
| 22:04.42 | Mon|A|rch | ChannelZ, thanks |
| 22:05.33 | ChannelZ | I think you mean func_uri and no that's something different |
| 22:05.52 | Mon|A|rch | ah |
| 22:05.55 | Mon|A|rch | looked like an L |
| 22:07.06 | Mon|A|rch | so i guess I should recompile >< |
| 22:07.19 | ChannelZ | Guess you can look at your config.log from your last build and see if any of the curl tests failed |
| 22:07.24 | igcewieling | go into menuconfig and verify the func_curl is stared |
| 22:07.38 | ChannelZ | Make sure you have libcurl-dev (or equivilant) installed |
| 22:07.58 | Mon|A|rch | checking if libcurl is installed now |
| 22:08.30 | igcewieling | Mon|A|rch: not libcurl-devel is the important one |
| 22:08.33 | Mon|A|rch | well, I've got libcurl installed |
| 22:08.59 | Mon|A|rch | ah |
| 22:09.03 | Mon|A|rch | didn't have libcurl-devel |
| 22:09.05 | igcewieling | Mon|A|rch: but that does not have the header files for compiling software using libcurl. this is why you need the -devel |
| 22:09.25 | Mon|A|rch | thanks |
| 22:11.48 | Mon|A|rch | well, libcurl-devel is installed, now here's hoping i don't foul up the recompile and ruin the system |
| 22:11.50 | Mon|A|rch | takes snapshot |
| 22:11.58 | ChannelZ | ./configure first |
| 22:12.13 | Mon|A|rch | yeah |
| 22:12.43 | Mon|A|rch | I'll need to find the source first |
| 22:13.22 | ChannelZ | I thought you said you were looking in menuconfig already.. |
| 22:14.17 | Mon|A|rch | i should have said logs of menuselect that the last guy working on this project made |
| 22:14.18 | Mon|A|rch | sorry |
| 22:15.00 | Mon|A|rch | and, menuselect |
| 22:15.09 | ChannelZ | ah. |
| 22:16.21 | Mon|A|rch | here's hoping it's in usr/src |
| 22:17.40 | Mon|A|rch | woo, it is |
| 22:20.03 | Mon|A|rch | so, ./configure, then i guess make, then make install |
| 22:20.40 | Mon|A|rch | I'll try make menuselect first |
| 22:21.18 | Mon|A|rch | func_curl has a star next to it |
| 22:21.28 | Mon|A|rch | guess that means there shouldn't be any problems if i compile now |
| 22:23.39 | *** join/#asterisk Tall-guy (Tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 22:23.42 | ChannelZ | Ideally |
| 22:28.25 | Mon|A|rch | looks like it compiled without any significant problems |
| 22:28.33 | Mon|A|rch | cut.so is apparently not compatible |
| 22:28.35 | Mon|A|rch | but w/e |
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| 22:29.52 | ChannelZ | Did this system get transferred as binaries from some other install before or something? |
| 22:30.15 | Mon|A|rch | cool, it works |
| 22:30.29 | Mon|A|rch | thanks very much ChannelZ |
| 22:30.44 | Mon|A|rch | that appears to be what's happened |
| 22:30.48 | Tall-guy | Goofy request: So, I've got a Cisco UC540 and a SIP Trunk provider who runs asterisk and I'm having no end of trouble getting it working. Is there anyone willing to "lend" me a sip trunk for 30 minutes to test authentication/registration? |
| 22:31.17 | Mon|A|rch | I was hired as a developer for a different project, then the guy doing the asterisk project quits, so I'm not fully up to speed on what he did |
| 22:31.51 | ChannelZ | Tall-guy: I can make you a peer to test |
| 22:32.04 | Tall-guy | Channelz: that'd be awesome dude.....can I pm u? |
| 22:32.09 | ChannelZ | yah |
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| 22:33.03 | pietro | hello |
| 22:34.20 | pietro | a little OT: I need to decrypt an SRTP flow (in a PCAP file, I have the negotiated key and the algorithm). someone know some tools that do this ? |
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| 23:01.26 | ChannelZ | maybe ANONYMOUS can lolz |
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