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| 00:50.31 | ukine | what's better quality, PCMU or GSM ? |
| 00:54.53 | thehar | mulaw |
| 00:56.00 | ukine | so pcmu? |
| 00:57.36 | ChannelZ | ulaw/alaw are similar and either are better than gsm |
| 01:02.52 | ukine | ty ChannelZ , thehar |
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| 02:49.11 | zigzapgabe | hey fellas, i am trying to edit the url in /etc/asterisk/vm_email.inc but for some reason the formatting is getting screwed up in vi and in nano |
| 02:49.13 | zigzapgabe | any ideas? |
| 02:51.18 | zigzapgabe | i am running version 1.4.21.2 |
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| 03:33.59 | sam_affable | hello |
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| 03:38.03 | UnixDev | good evening all |
| 03:38.23 | UnixDev | how can I change the "Server: " sip header that asterisk writes when it generates sip messages? |
| 03:38.35 | UnixDev | i have tried setting useragent in the config to no avail |
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| 03:54.06 | joeyjones | ChannelZ: is 32 kbit pcma a bad codec? |
| 03:54.51 | joeyjones | maybe it was 64 kbit |
| 03:56.36 | xSmurf | this realtime ldap stuff is really shitty :( I think I'm just gonna generate some extensions in freepbx's tables and reload |
| 03:56.41 | xSmurf | blarg, fugly |
| 03:58.50 | p3nguin | joeyjones: ulaw/alaw are going to provide you with good sound quality at the expense of more bandwidth. |
| 03:59.24 | *** join/#asterisk talntid (~Eric@173.160.174.126) |
| 04:00.18 | talntid | In asterisk, it's trying to send packets to 192.168.1.141, which is the localip on my network for my polycom phone, but it's not on the same network as the asterisk box... i forget what the procedure is to resolve this... |
| 04:01.31 | joeyjones | p3nguin: i do have some issues with call delay as it is though |
| 04:01.36 | joeyjones | >123ms |
| 04:01.39 | p3nguin | talntid: Properly configure the system for use with NAT: nat, localnet, externhost or externip, and canreinvite |
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| 04:07.46 | talntid | I think the issue is just that it is because I am behind a NAT, with my stuff, away from the pbx.... ? |
| 04:08.25 | talntid | the pbx isn't behind NAT |
| 04:09.16 | p3nguin | talntid: Perhaps. If Asterisk is behind NAT, it has to be configured accordingly. Likewise, if the phones are behind a different NAT, the same is still true but for the phones. |
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| 04:09.38 | chaimf | hi everyone |
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| 04:11.00 | p3nguin | talntid: Often, the only thing needed is to configure nat=yes and canreinvite=no in the peer definition for the phones which are behind a NAT. |
| 04:13.06 | talntid | i know this isn't an ideal config, but i'll clean up the passwords and stuff when it works... |
| 04:13.08 | talntid | http://pastebin.com/FwupSugr |
| 04:13.17 | talntid | that's my current sip.conf. |
| 04:13.36 | p3nguin | You've disabled all codecs. |
| 04:13.45 | talntid | if i place a call with one of the externaladmin phones, it goes through, ..... SHIT |
| 04:14.03 | talntid | wait, they are allowed in the externaladmin definition |
| 04:14.40 | p3nguin | I noticed that. |
| 04:15.05 | talntid | the calls go through, outgoing... the target cell phone rings, but no audio either way. |
| 04:16.20 | talntid | pbx is not natted, I can connect directly to it.... phone is a polycom ip550, it's connected to the pbx, but pbx still trying to send packets to the IP address of the polycom on my local network. hmm.. |
| 04:16.23 | p3nguin | No audio indicates a problem with RTP. I hate those problems because there isn't much that can be done to try to fix them. |
| 04:16.57 | talntid | under "sip show peers", Nat = N |
| 04:17.04 | p3nguin | That's good. |
| 04:17.18 | joeyjones | talntid: try setting nat=1 insecure=invite,port for the extension in sip.conf |
| 04:17.38 | p3nguin | joeyjones: Extensions are not found in sip.conf. |
| 04:17.46 | joeyjones | w/e |
| 04:17.53 | joeyjones | the sip info for it |
| 04:18.19 | p3nguin | talntid: Anyway... did you make sure the phone doesn't have NAT traversal enabled? |
| 04:19.08 | talntid | i'm not sure I have ever seen that option on polycom.. |
| 04:19.44 | p3nguin | I don't use Polycom phones, so I didn't know. |
| 04:19.48 | chaimf | talntid, what router are you using? |
| 04:19.57 | chaimf | talntid, both sides nat? or only on the phone side? |
| 04:20.03 | chaimf | rtp debug shows what? |
| 04:20.27 | p3nguin | I ran into this problem a while back. The solution was to change the router. |
| 04:21.11 | talntid | chaimf, only phone side. |
| 04:21.31 | talntid | chaimf, router is a DD-WRT enabled WRT54g, on phone side |
| 04:21.35 | p3nguin | rtp debug shows the rfc1918 address of the phone, but sip debug shows the outside address. It allowed for proper call establishment, but the media had no path. |
| 04:22.25 | talntid | http://pastebin.com/0Vu3xH2S |
| 04:22.27 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 04:22.29 | talntid | that's the rtp debug |
| 04:22.48 | talntid | my ip, on the phone side: 173.160.174.126 |
| 04:23.25 | chaimf | so you are hearing on the asterisk side just not on the phone side? |
| 04:23.36 | talntid | no audio, either way |
| 04:23.39 | p3nguin | I don't know if I've ever seen anyone in here using dd-wrt with VoIP successfully. There's always a complaint about something not working with it. |
| 04:23.52 | chaimf | according to the rtp output you get rtp packets |
| 04:24.38 | shmaltz | talntid, where is the other phone? |
| 04:24.50 | talntid | shmaltz, at&t's network. |
| 04:25.06 | talntid | http://pastebin.com/0hHMB9Ud |
| 04:25.08 | shmaltz | canreinvite is set to no? |
| 04:25.11 | talntid | that's the sip debug |
| 04:25.28 | talntid | http://pastebin.com/FwupSugr that's my current sip.conf. |
| 04:26.04 | talntid | the phone in question, is "ericand" |
| 04:26.16 | p3nguin | What is at IP address 10.21.0.22? |
| 04:26.24 | talntid | the asterisk server |
| 04:26.39 | p3nguin | That's an rfc1918 address, but you say it isn't behind NAT. |
| 04:26.45 | talntid | er |
| 04:27.05 | shmaltz | thanks p3nguin for saying what I thought |
| 04:27.40 | *** join/#asterisk hellome (~hellome@c-69-253-65-8.hsd1.pa.comcast.net) |
| 04:27.44 | talntid | isn't it considered out from behind the nat, when i can freely access it from anywhere? |
| 04:27.55 | p3nguin | no |
| 04:27.58 | talntid | no port forwarding and all |
| 04:28.06 | p3nguin | 1-to-1 NAT |
| 04:28.07 | shmaltz | what? |
| 04:28.20 | shmaltz | externip and what not has to be set |
| 04:28.24 | p3nguin | yep |
| 04:28.38 | p3nguin | If not 1-to-1, then you've set it up in DMZ. |
| 04:28.43 | shmaltz | thats why I saw the NO NAT in the sip debug output |
| 04:28.59 | talntid | it's 1-to-1 |
| 04:29.15 | talntid | which i thought, means it disables nat, effectively? |
| 04:29.21 | shmaltz | talntid, it dosnt matter if its 1to 1 its still nat |
| 04:29.24 | p3nguin | nat, localnet, externip or externhost, and canreinvite must be set accordingly on Asterisk. |
| 04:29.51 | talntid | roger that. |
| 04:30.22 | shmaltz | case closed, next :D |
| 04:30.40 | p3nguin | Not closed yet. |
| 04:31.30 | p3nguin | There's always a possibility it will still not work after those settings are changed. |
| 04:31.58 | shmaltz | with this type of misconfiguration its a new case :P |
| 04:32.03 | p3nguin | heh |
| 04:32.18 | p3nguin | Let's be optimistic. |
| 04:32.27 | shmaltz | and talntid, dont give up, one never learns if they don't ask |
| 04:32.42 | shmaltz | once upon a time I was the one asking |
| 04:33.58 | talntid | i have been working on this for about 3 days, before i came here |
| 04:34.09 | talntid | it's kicking my ass. so i am asking for help. heh :) |
| 04:34.33 | talntid | i have a fully functional 25 person call center running on this, but never needed to work outside... trying to get an outside agent working. :) |
| 04:34.54 | talntid | can't afford to put a dedicated asterisk box at their site, like would be proper... |
| 04:35.03 | p3nguin | You could always set up VPN for it. |
| 04:35.10 | joeyjones | p3nguin: how can i tell if my asterisk install supports g722 64kbit? |
| 04:35.17 | shmaltz | talntid, shouldn't need to, just configuring nat should do the trick |
| 04:35.36 | shmaltz | show codecs still works? |
| 04:35.36 | shmaltz | joeyjones, what version? |
| 04:35.38 | sam_affable | hey |
| 04:35.39 | p3nguin | joeyjones: core show translation recalc 10 |
| 04:35.42 | joeyjones | 1.4 iirc |
| 04:36.18 | shmaltz | yeah, 1.4 supports show codecs |
| 04:36.25 | shmaltz | just type in show codecs in cli |
| 04:36.32 | talntid | a VPN would be useful, but the outside agent needs to have a way to connect to the VPN. that adds an appliance, which... is the same cost as a linux box |
| 04:36.35 | p3nguin | core show codecs doesn't show what your system supports. |
| 04:36.38 | sam_affable | asterisk1.8+h323plus 1.20.0 anyone?? asterisk menuselect wont recognize h323plus |
| 04:37.13 | shmaltz | p3nguin you are right sorry |
| 04:37.22 | shmaltz | joeyjones show translation |
| 04:37.32 | p3nguin | talntid: dd-wrt doesn't work as a VPN endpoint? |
| 04:38.16 | talntid | i'm merely testing this here at my house |
| 04:38.21 | p3nguin | oh |
| 04:38.24 | talntid | before fumbling with it for 3 days, at the remote agent |
| 04:38.26 | talntid | ;) |
| 04:38.43 | joeyjones | talntid: what abouyt trying a pc softphone to see if it could be the phone itself causing sisues? |
| 04:38.48 | joeyjones | *issues |
| 04:38.54 | talntid | using my android, using sipdroid |
| 04:38.57 | talntid | and my polycom phone |
| 04:39.02 | talntid | both fail |
| 04:39.10 | joeyjones | talntid: sipdroid is a bitch, it can randomly fail |
| 04:39.16 | p3nguin | After making the changes for NAT, still no luck? |
| 04:39.18 | talntid | added the options i was instructed to... same issue currently :) |
| 04:39.28 | talntid | well, same result. maybe not same issue |
| 04:39.45 | p3nguin | I guess we need a new sip debug and rtp debug. |
| 04:39.58 | joeyjones | talntid: pastebin from sip.conf the general section and this particular phone's section\ |
| 04:40.21 | talntid | http://pastebin.com/vVgGSWiU |
| 04:40.49 | talntid | http://pastebin.com/fBXY6RxT |
| 04:40.55 | talntid | the phone is "ericand" |
| 04:41.18 | p3nguin | # |
| 04:41.19 | p3nguin | extenip=66.208.251.171 |
| 04:41.22 | p3nguin | type |
| 04:41.25 | p3nguin | typo, rather |
| 04:41.29 | p3nguin | externip |
| 04:42.24 | p3nguin | Does nat=1 even work? I thought valid values were yes, no, route, and never. |
| 04:42.25 | talntid | genius ;) |
| 04:42.39 | joeyjones | p3nguin: nat=1 does work |
| 04:42.57 | joeyjones | and is 2 less chars :p |
| 04:43.56 | talntid | ok, now it works.. |
| 04:43.59 | talntid | except, one way audio |
| 04:44.10 | joeyjones | talntid: do you have a test extension setup? |
| 04:44.14 | shmaltz | talntid, which way? |
| 04:44.20 | shmaltz | to asterisk or from asterisk? |
| 04:44.41 | talntid | when i speak into my voip phone, it goes out and i can hear it on the other end. |
| 04:44.59 | joeyjones | talntid: setup an extension to play something and one for echo |
| 04:45.00 | talntid | but when i speak into the other end, it does not come over the voip phone speaker |
| 04:45.12 | talntid | ok |
| 04:45.23 | joeyjones | talntid: like http://pastebin.com/6FagTXdx |
| 04:45.38 | joeyjones | 500 plays weasels and 600 pechos |
| 04:45.52 | shmaltz | talntid, but asterisk is sending it |
| 04:45.56 | p3nguin | Don't forget to convert your 1.2 dialplan pipe lines to commas. |
| 04:46.07 | shmaltz | talntid, the firewall on the other end seems to block it |
| 04:46.22 | shmaltz | p3nguin, whats that? |
| 04:46.29 | talntid | this is a fresh dialplan |
| 04:46.32 | joeyjones | i wonder if sipdroid supports ulaw/alaw... |
| 04:46.44 | p3nguin | exten => 500,1,Verbose(1|Echo test application) <--- from the pastebin |
| 04:46.51 | shmaltz | p3nguin, no more | for option seperations in 1.4? |
| 04:47.09 | p3nguin | Changed to commas after 1.2. |
| 04:47.17 | shmaltz | only started using 1.4 1 week ago |
| 04:47.33 | shmaltz | is checking for | in the 2 updated boxes |
| 04:48.14 | talntid | testing :) |
| 04:48.16 | shmaltz | p3nguin, voicemail.conf could still take |? |
| 04:48.17 | p3nguin | I've used pipe lines in my earlier 1.4 boxes and it did work, but I don't know if it still does in current 1.4 versions. |
| 04:48.41 | p3nguin | s/current/more recent/ |
| 04:48.45 | joeyjones | i guess i should get my internal actually calling other extensions... |
| 04:49.21 | shmaltz | p3nguin, what about gotoiftime command? |
| 04:50.09 | p3nguin | exten => s,n,GotoIfTime(9:00-18:00,mon-fri,*,*?daytime,s,1) |
| 04:50.27 | talntid | hmm =] |
| 04:50.32 | talntid | it would appear.. to work =] |
| 04:50.39 | talntid | anyone care to test, by registering a phone to it? |
| 04:51.01 | p3nguin | I could register my asterisk system to it, I suppose. |
| 04:51.01 | joeyjones | talntid: call 500/600? |
| 04:51.02 | talntid | and placing a call. to ensure it will work when i go to someone elses place to install it. I'll be driving 2 hours to install this.. |
| 04:51.21 | talntid | joey, the echo test works, and i just called a friend. |
| 04:51.33 | joeyjones | talntid: i'm connected to your already :p |
| 04:51.35 | talntid | on the polycom |
| 04:51.40 | talntid | sweet =D |
| 04:51.42 | joeyjones | *yours |
| 04:51.45 | talntid | echo test: 111 |
| 04:51.50 | talntid | weasels: 6000 |
| 04:52.13 | talntid | and it will make outgoing calls. |
| 04:52.18 | talntid | feel free |
| 04:52.20 | talntid | ;) |
| 04:52.26 | joeyjones | talntid: lol |
| 04:52.35 | joeyjones | you may want to change passes :p |
| 04:52.41 | talntid | definately will ;) |
| 04:52.47 | joeyjones | or you'll wind up with a large long distance bill :p |
| 04:52.59 | talntid | cool thing is, i pay $830/mo for unlimited |
| 04:53.09 | talntid | but, 911 calls would suck. |
| 04:53.16 | p3nguin | holy cow, that's a lot of money! |
| 04:53.29 | talntid | i put 230,000 minutes through it monthly |
| 04:54.24 | shmaltz | talntid, what type of connection? |
| 04:54.29 | talntid | it's a t1 PRI |
| 04:54.41 | shmaltz | so unlimited on all 23 channels? |
| 04:54.45 | talntid | yes |
| 04:55.05 | shmaltz | talntid, optimum cable is cheaper 23*$30 |
| 04:55.22 | talntid | I have two comcast cable connections, which are very fast |
| 04:55.24 | joeyjones | is thinking of setting up a cheap longdistance calling service, using prepaid per minute service through rapidvox |
| 04:55.25 | shmaltz | talntid, you have optimum in your area? |
| 04:55.44 | talntid | but, the quality leaves a bit to be desired, on a lot of the PPU providers.. |
| 04:55.56 | talntid | i have been testing flowroute the last few days |
| 04:55.57 | shmaltz | optimum started offering PRI handoff for $30 a channel unlimited |
| 04:56.14 | talntid | honestly, i'd pay $5000/month if needed. |
| 04:56.27 | talntid | just, need the calls to go through, reliably.. every time |
| 04:56.52 | joeyjones | talntid: call center or re-selling? |
| 04:57.23 | talntid | call center |
| 04:57.27 | talntid | www.rtui.com |
| 04:58.03 | joeyjones | talntid: mind if i pm? |
| 04:58.39 | talntid | go |
| 04:59.26 | talntid | go for it :) |
| 05:10.21 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 05:24.31 | joeyjones | p3nguin: how can i tell which codec a peer is using for a call? |
| 05:25.24 | p3nguin | sip show channels should show it for sip calls. |
| 05:26.44 | p3nguin | Also, sip show channel <your channel id> |
| 05:28.54 | joeyjones | p3nguin: damn, looks like the phone is using pcma 64kbit |
| 05:29.16 | p3nguin | As configured, I presume. |
| 05:30.47 | joeyjones | p3nguin: i'm guessing that the sip.conf allow/disallow won't affect the format that the phone uses... |
| 05:31.02 | p3nguin | Actually, it directly affects it. |
| 05:31.20 | p3nguin | disallow=all allow=ulaw <-- this allows only ulaw to be used on the phone |
| 05:31.43 | p3nguin | What codec do you want the phone to use? |
| 05:31.53 | joeyjones | ulaw |
| 05:32.17 | joeyjones | i have disallow=all allow=ulaw under general in sip.conf |
| 05:32.27 | p3nguin | Change or set it in the peer definition for that phone. |
| 05:33.34 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 05:34.34 | joeyjones | p3nguin: now the phone shows PCMU... |
| 05:34.40 | joeyjones | mayb e PCMA was alaw |
| 05:34.43 | p3nguin | it is |
| 05:34.57 | p3nguin | pcma is alaw, pcmu is ulaw (mu-law) |
| 05:35.23 | p3nguin | Both are G.711 64k |
| 05:35.29 | joeyjones | is g722 better than ulaw? |
| 05:35.56 | p3nguin | g722 is wideband, so it would sound better if you have the capability to use it end-to-end. |
| 05:37.14 | joeyjones | if i set allow=g722 allow=ulaw would g722 be tried first? |
| 05:37.36 | p3nguin | Probably, yes. |
| 05:38.53 | joeyjones | weird, g722 gives me a lot less delay in the call when using echo |
| 05:41.35 | ChannelZ | Well depending on the quality, the packets are smaller. |
| 05:44.15 | joeyjones | it seems though that g722 to pbx and ulaw to sip provider fails |
| 05:44.37 | p3nguin | Pastebin the sip debug. |
| 05:45.55 | joeyjones | http://pastebin.com/e0dAmKP8 is what i saw in debug |
| 05:46.15 | joeyjones | w/ 1000 as g722, ulaw and rapidvox as ulaw |
| 05:46.51 | p3nguin | "No audio format found to offer." <-- this is the result of the part you didn't include in the paste. |
| 05:47.18 | p3nguin | The part you didn't include is the part I was interested in. |
| 05:49.41 | joeyjones | p3nguin: maybe http://pastebin.com/eTcJ6d4h |
| 05:49.51 | joeyjones | that was all my CLI would hold |
| 05:50.31 | p3nguin | Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x100c (ulaw|alaw|g722)/video=0x0 (nothing), combined - 0x1004 (ulaw|g722) |
| 05:51.27 | p3nguin | This says the peer supports both ulaw and g722, so I don't see a reason no codec could be agreed on. |
| 05:51.34 | joeyjones | exactly |
| 05:51.46 | joeyjones | weird |
| 05:51.48 | jsolares | maybe the peer doesn't support g722 |
| 05:52.04 | joeyjones | i was able to make a call to echo 2/ g722 |
| 05:52.09 | joeyjones | *w/ |
| 05:52.20 | joeyjones | and rapidvox xupports ulaw |
| 05:52.23 | joeyjones | *supports |
| 05:53.01 | jsolares | but does rapidvox support g722? |
| 05:53.16 | p3nguin | Is that debug not for the call between Asterisk and rapidvox? |
| 05:53.32 | p3nguin | sip set debug peer rapidvox ? |
| 05:53.44 | jsolares | it seems the first is for phone <-> asterisk, and then rapidvox has no sip debug |
| 05:54.04 | ChannelZ | "Rapidvox supports the use of G.711, GSM, G.729 and iLBC audio codecs." |
| 05:54.25 | p3nguin | G.711 could be ulaw or alaw, but does it have to mean both? |
| 05:55.30 | p3nguin | If we see the debug between rapidvox and asterisk, we'll know exactly what it supports. I assumed that's what I was looking at before. |
| 05:55.31 | ChannelZ | Not necessarily, I suppose. |
| 05:55.45 | joeyjones | apparently asterisk 1.4 doesn;t support transcoding g722 |
| 05:56.00 | p3nguin | Why wouldn't it? |
| 05:56.26 | jsolares | nothing a core show translation wouldn't show |
| 06:01.06 | p3nguin | joeyjones: http://users.netplex.net/~andrew/asterisk/ |
| 06:09.54 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-pihvcjvljopevypn) |
| 06:11.51 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ioykawxjkmoikdvu) |
| 06:12.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cygktambsxrexyif) |
| 06:13.32 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-awutrnfjyvbixzff) |
| 06:14.34 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:20.24 | p3nguin | joeyjones: I tried the patch with version 1.4.37, and it all ended with failure. |
| 06:21.20 | p3nguin | http://pastebin.com/dcqK2Lj6 |
| 06:21.50 | p3nguin | I'll have to remember tomorrow to check that patch and see if I can figure out why it doesn't work. |
| 06:21.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:22.27 | joeyjones | p3nguin: i'm on Asterisk 1.4.21.2~dfsg-3+lenny1 built by pbuilder @ grnetbox on a x86_64 running Linux on 2009-12-14 19:04:56 UTC |
| 06:58.01 | *** join/#asterisk Leddy (leddy@aqua.surgehost.net) |
| 07:00.45 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 07:06.02 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 07:14.24 | *** join/#asterisk AndyRomano (~Adium@195.34.154.20) |
| 07:20.06 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
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| 07:29.35 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:51.55 | joeyjones | hhmm... |
| 07:58.01 | joeyjones | hhmm... |
| 07:58.28 | joeyjones | i wonder what the best way is to get decent call quality when using 3G and getting > 325ms delay with ulaw/gsm |
| 07:59.02 | p3nguin | You'll never get good quality using gsm codec. |
| 07:59.09 | p3nguin | It's an oxymoron. |
| 07:59.37 | joeyjones | true |
| 07:59.40 | joeyjones | well |
| 07:59.44 | joeyjones | quality/delay |
| 08:00.01 | p3nguin | g729 might be worth looking at. |
| 08:00.29 | joeyjones | p3nguin: no client support |
| 08:01.01 | joeyjones | g722, silk24, silk16, silk8, pcma, pcmu, speex, gsm, bv16 |
| 08:04.49 | joeyjones | p3nguin: well, it seems like it's time to actually setup a dialplan :p |
| 08:06.12 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:11.21 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 08:11.24 | DND | hi guys |
| 08:11.30 | DND | i need help with chanspy |
| 08:11.47 | DND | currently its configured to have a password then let the user have to enter the extension |
| 08:12.19 | DND | now what i wanted to do is. i wanted to restrict some extensions since some being used by managers, finance etc. |
| 08:12.30 | DND | how can i restrict it to specific people only? |
| 08:13.02 | kaldemar | give the people their own codes to authenticate. |
| 08:13.18 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:13.33 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:13.55 | DND | hmm houw about groups? |
| 08:14.03 | kaldemar | if you want to restrict it to certain phones, use some information that is related to the devices for restrictions. |
| 08:14.13 | kaldemar | what do you mean by groups? |
| 08:14.45 | DND | i mean i will create like "agent" group then i will code asterisk chanspy to listen on that group extension |
| 08:15.33 | DND | something like this: http://www.jonathanmanning.com/2009/10/29/monitoring-agents-in-asterisk-with-chanspy/ |
| 08:15.36 | DND | check example 3 |
| 08:15.47 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 08:16.49 | DND | im not sure what the [default] context is for |
| 08:16.56 | DND | musta pinoyskull |
| 08:17.16 | DND | (how are you) |
| 08:20.28 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 08:20.32 | giany | Hello |
| 08:20.51 | kaldemar | the default context is e.g. for unauthenticated calls if no other context is defined and also used in auto fall-through. i'd suggest keeping it empty or removing it. |
| 08:21.45 | giany | I have this problem with a provider that has different ips for SIP signalign and RTP signaling. Problem is that I don't hear the ringing sound on the phone that calls the Asterisk Box. Is there any setting regarding this? |
| 08:23.42 | kaldemar | DND: but in that example, g() option is used to determine a group that is spied on. when the calls come in, they are added to the group before going into the queue. |
| 08:23.55 | *** join/#asterisk Corydon76-home (green@c-69-137-80-31.hsd1.tn.comcast.net) |
| 08:23.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:24.35 | kaldemar | giany: your phone is supposed to generate the ringing sound unless the call is already answered. |
| 08:25.41 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:25.44 | DND | hmm |
| 08:27.40 | DND | or should i just do this: exten => _88XXXX,1,Chanspy(SIP/${EXTEN:2}|b) |
| 08:27.55 | DND | i will do it per extension |
| 08:28.31 | DND | so will just change XXXX to the ext number then change to Chanspy(SIP/extension) |
| 08:28.40 | DND | we just need 5 |
| 08:28.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:29.10 | DND | but will also add password protection |
| 08:30.01 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 08:30.36 | nicola_pav | hello. asterisk crashed with the following error: asterisk[25796]: segfault at c ip b7f71450 sp b282873c error 4 in libpthread-2.8.so[b7f6a000+14000] |
| 08:30.44 | nicola_pav | any hints? |
| 08:30.58 | nicola_pav | tired to google but no concrete solutions |
| 08:31.02 | DND | check line 25796 in /var/log/asterisk/full |
| 08:31.25 | nicola_pav | DND: i will now |
| 08:31.29 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 08:33.40 | *** join/#asterisk Tim_Toady (~moi@188.4.87.53.dsl.dyn.forthnet.gr) |
| 08:33.57 | kaldemar | nicola_pav: try to reproduce the crash and see this: http://svn.digium.com/svn/asterisk/tags/1.8.0/doc/backtrace.txt |
| 08:36.59 | nicola_pav | DND: nothing abnormal there |
| 08:37.07 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 08:37.09 | nicola_pav | kaldemar: i will try to reporduce |
| 08:37.20 | giany | kaldemar: on the extension where I call I have sometransfer rules..when those numbers from the rules are called I don't hear the ringing. |
| 08:41.23 | nicola_pav | kaldemar: thanks for the link, its definitely helpful |
| 08:42.18 | nicola_pav | kaldemar: i have asterisk 1.4 |
| 08:42.32 | nicola_pav | i can see the option g |
| 08:42.45 | nicola_pav | but does it work as supposed? |
| 08:44.32 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:45.01 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
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| 08:48.08 | *** join/#asterisk SaS1 (~Adium@ip-87-86-154-70.easynet.co.uk) |
| 08:48.21 | kaldemar | nicola_pav: yes it does. |
| 08:48.59 | kaldemar | giany: what kind of transfer rules? |
| 09:00.12 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
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| 09:10.07 | *** part/#asterisk Kryo (~Adium@ip-87-86-154-70.easynet.co.uk) |
| 09:11.22 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
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| 09:18.42 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 09:19.03 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 09:22.39 | *** join/#asterisk friendly12345 (~friendly@ppp118-209-88-20.lns20.mel4.internode.on.net) |
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| 09:51.06 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 09:52.32 | verywiseman | i follow instructions in http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO , but when i press *0 , nothing happen, would you help me? |
| 09:55.49 | verywiseman | forget what i write previously |
| 09:58.21 | tzafrir | What was the problem? |
| 10:03.55 | verywiseman | tzafrir, in http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO . what is that meaning ? exten => s,n,Dial(SIP/${DEST}@dynamic-nway-dest,,g) |
| 10:04.38 | tzafrir | verywiseman, those lines should be /etc/asterisk/extensions.conf |
| 10:04.42 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 10:05.09 | verywiseman | tzafrir, yes , i know |
| 10:06.53 | verywiseman | tzafrir, look, i am in exten 1111 , and i called 2222 , but when i press *0 , 1111 is appearing disconnected in 2222 ,why? |
| 10:08.58 | tzafrir | how can you tell it is disconnected? |
| 10:12.45 | verywiseman | tzafrir, in ext 2222 , in ip phone screen , 1111 is disconnected |
| 10:12.46 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:16.41 | *** join/#asterisk jsjc (~jsjc@42.Red-88-22-178.staticIP.rima-tde.net) |
| 10:17.24 | jsjc | Hi! I have a little issue, when my PPOE DSL connection drops out and reconnects the sip registry appears as registered but cannot get any incoming stuff going trough... |
| 10:22.46 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:23.45 | *** join/#asterisk riscphree (~riscphree@h173.113.90.75.dynamic.ip.windstream.net) |
| 10:27.30 | *** part/#asterisk friendly12345 (~friendly@ppp118-209-88-20.lns20.mel4.internode.on.net) |
| 10:36.13 | *** join/#asterisk guilhermebr (~Guilherme@189.50.119.253) |
| 10:47.24 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 10:47.48 | verywiseman | in http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO , what is "exten => 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv) " meaning? |
| 10:52.28 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 10:57.45 | kaldemar | verywiseman: it sets nway-inv and nway-noinv as activated dynamic features. they're defined in [applicationmap] of features.conf. |
| 10:58.14 | eduzimrs | anyone knows about this problem ? > chan_sip.c:15896 handle_request_subscribe: SUBSCRIBE failure: no Accept header: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'hints', subscribeuri: '' |
| 11:01.36 | kaldemar | eduzimrs: looks like asterisk wants an Accept header in SUBSCRIBE messages and there is none. |
| 11:03.08 | eduzimrs | kaldemar its expecting something that doesn´t was set? is that? |
| 11:03.26 | kaldemar | what? |
| 11:03.31 | eduzimrs | kaldemar ops, sorry |
| 11:04.05 | eduzimrs | kaldemar * is expecting something that wasn`t set? is that? |
| 11:05.29 | kaldemar | the warning says so. what client are you using? |
| 11:05.57 | eduzimrs | kaldemar pangolin |
| 11:06.48 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:07.14 | kaldemar | is that causing you problems? |
| 11:09.32 | eduzimrs | kaldemar im monitoring this sip client in a TI phone, and when i unregister this peer the light doesnt turn red as supose to be |
| 11:09.52 | eduzimrs | kaldemar this WARINING appear when a register this peers |
| 11:10.52 | dimm | i see usb-3g-modem in server with asterisk, i know that this device was use with asterisk. How i can find something in asterisk's config related to usb-3g-modem ? |
| 11:11.54 | WIMPy | dimm: IIRC datacard.conf |
| 11:11.59 | *** join/#asterisk coppice (~chatzilla@m121-202-79-28.smartone-vodafone.com) |
| 11:13.41 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 11:19.30 | kaldemar | eduzimrs: well, asterisk requires the accept header so your pretty much out of luck with that if you can't make the client send it. of course you could modify chan_sip yourself to fix it. |
| 11:25.01 | dimm | WIMPy, i look in datacard.conf, but see nothing, can you say some words about it - http://paste.org.ru/?zpfr2z ? |
| 11:27.14 | *** part/#asterisk SaS1 (~Adium@ip-87-86-154-70.easynet.co.uk) |
| 11:29.21 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:32.31 | WIMPy | dimm: That sure looks funny. The two devices wiuld usually be /dev/ttyUSBx. |
| 11:32.36 | WIMPy | Maybe someone renamed them? |
| 11:33.23 | dimm | WIMPy, http://paste.org.ru/?ii5n7w |
| 11:33.28 | WIMPy | And setting gain on a digital connection doesn't seem to obvious to me, either. |
| 11:35.46 | dimm | WIMPy, http://paste.org.ru/?g2grel |
| 11:36.05 | dimm | how i can found what device is using for data, and what using for voice? |
| 11:36.18 | dimm | s/device/tty/ |
| 11:36.58 | *** join/#asterisk SaS1 (~Adium@ip-87-86-154-70.easynet.co.uk) |
| 11:37.03 | dimm | how i can found what tty is using for AT-commands, and what using for audio-connection? |
| 11:37.38 | dimm | http://paste.org.ru/?ljza1x |
| 11:38.21 | WIMPy | You have to know. Or try. |
| 11:38.44 | dimm | :) |
| 11:38.49 | WIMPy | The first device seems to be for commands usually. |
| 11:39.10 | dimm | ls -l showing symlinks from /dev/data1 to /dev/ttyUSB2 |
| 11:39.26 | WIMPy | But it also depends on your kernel and maybe the modeswitch util. |
| 11:41.36 | WIMPy | I have an E1550 that used to show as 4 devices. Now it's only 3. But unfortunately the voice functionality is disabled :-( |
| 11:42.44 | dimm | also periodically i see in asterisk's cli that: |
| 11:42.50 | dimm | -- Datacard 1 trying to connect on /dev/data1... |
| 11:43.10 | dimm | now i know, that this is about 3g modem, am i right? |
| 11:43.26 | WIMPy | It is |
| 11:45.51 | dimm | ok, i re-connect usb-device to usb-port and look at 'tail -n 0 -f /var/log/messages' and in 'dmesg' output |
| 11:46.03 | dimm | i see thatdevice recognized correct |
| 11:46.18 | dimm | how i can "connect " this device to asterisk? |
| 11:46.42 | dimm | CLI> datacard show devices |
| 11:46.56 | dimm | is return that device 1 is not connected |
| 11:51.01 | WIMPy | Well I could never use it bacause I don't have the right stick, but I'm pretty sure it showed as connected, even if it wasn't usable. |
| 11:58.58 | *** join/#asterisk secesh (~secesh@adsl-074-167-019-138.sip.sav.bellsouth.net) |
| 12:05.05 | tzafrir | WIMPy, if you have ttyUSB2, the mode has already switched |
| 12:05.10 | *** join/#asterisk anebi (~anebi@91.207.191.17) |
| 12:05.29 | tzafrir | Also: ls -l /dev/serial/by-id will give you a slightly clearer picture |
| 12:05.55 | anebi | hi, i installed asterisk 1.6 with freepbx on one of our machines and i have an issue with choppy audio when i dial to echo test, voicemail, etc. |
| 12:06.22 | tzafrir | WIMPy, some (many? most?) modems provide more than a single tty interface. In the ones I encountered, just use the last of them |
| 12:06.29 | anebi | right now asterisk uses wav sound files and i want to change this as i want to use gsm or other formats, but not wave |
| 12:06.53 | tzafrir | anebi, what's your problem with .wav ? |
| 12:07.04 | anebi | wav*. i cannot find where this is defined in configuration files and i need your help. |
| 12:07.27 | tzafrir | Asterisk basically tries to play the sound file of "the best format" |
| 12:07.28 | anebi | tzafrir: the audio is choppy, i thought that if i change it to gsm or other format will be better then wav |
| 12:07.49 | tzafrir | for local files: no |
| 12:08.17 | tzafrir | What codec do you use for the voip link? |
| 12:08.43 | coppice | tzafrir: do you know why they have multiple ttys? |
| 12:09.33 | tzafrir | Not really sure. I faintly recall an article by the NetworkManager author about this |
| 12:09.34 | anebi | tzafrir: we have this order of codecs set on phones: G. 711 (u-law), G. 711 (a-law), G. 729A, GSM FR, G. 722, G. 726 (32 kbits/s) |
| 12:10.00 | tzafrir | anebi, are the phones in your LAN? |
| 12:10.47 | anebi | <PROTECTED> |
| 12:16.33 | k-man | any idea what the functionality differences are between the siemens 580IP and the 470IP? |
| 12:16.47 | *** join/#asterisk msetim (~setim@187.112.62.185) |
| 12:16.51 | msetim | hello guys |
| 12:17.08 | msetim | is secure use cdr uniqueid like an primary key at database? |
| 12:20.50 | ectospasm | msetim: afaik, the uniqueid should be completely unique amongst all CDR entries, kinda like a primary key in a database. The CDR being the database in this case |
| 12:22.09 | msetim | ectospasm, thanks :) it's applied to queue_log... but is a compost key (uniqueid, event) |
| 12:24.41 | *** join/#asterisk Stratisphere (~Stratisph@unaffiliated/stratisphere) |
| 12:24.47 | Stratisphere | hey all |
| 12:24.58 | Stratisphere | anyone familiar with the LDK 100 PBX? |
| 12:25.17 | Stratisphere | by LG :P |
| 12:28.23 | tzafrir | anebi, problem with timing? |
| 12:30.23 | anebi | tzafrir: it is possible. the server is on amazon ec2 instance and i cannot change kernel settings there as someone suggested to use these kernel params: acpi=off noapic nosmp nolapic clock=pit. but i cannot do this on ec2. i need to find another solution |
| 12:31.05 | tzafrir | anebi, what version of dahdi? What kernel version? |
| 12:31.15 | DeHackEd | EC2 uses Xen. So it probably has no acpi, apic or even pit clock |
| 12:31.17 | tzafrir | Sorry: s/dahdi/asterisk/ |
| 12:31.28 | *** join/#asterisk domedan (~domedan@c83-253-183-67.bredband.comhem.se) |
| 12:33.55 | anebi | tzafrir: kernel: 2.6.18-xenU-ec2-v1.0, asterisk: 1.6.0.5, dahdi: dahdi-linux 2.1.0.4 dahdi-tools 2.1.0.2 |
| 12:34.11 | anebi | DeHackEd: it is possible, i haven't check this yet |
| 12:34.33 | DeHackEd | check if the file /sys/hypervisor/type exists |
| 12:35.11 | anebi | DeHackEd: yes, with content "xen" |
| 12:35.23 | DeHackEd | then it's probably PV |
| 12:35.43 | DeHackEd | xen services as the clock source, depending on the version of xen there are timing quirks |
| 12:36.29 | tzafrir | anebi, asterisk >= 1.6.1 has support for an external timing source. With your version you only have dahdi |
| 12:36.56 | tzafrir | What's the output of: dahdi_test -v -c 6 #? |
| 12:38.11 | anebi | DeHackEd: i see. tzafrir: here is the result: http://pastebin.com/eTgDCAui |
| 12:38.52 | tzafrir | anebi, hmm... not so good, indeed |
| 12:38.58 | tzafrir | Try a newer version of dahdi |
| 12:39.18 | tzafrir | IIRC 2.3.0 was a useful improvement there |
| 12:40.27 | anebi | tzafrir: hm, ok. i will try with a newer version. thanks a lot for trying to help me to all :) |
| 12:42.05 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 12:46.32 | *** join/#asterisk skrusty (~skrusty@83.166.176.39) |
| 12:46.43 | skrusty | afternoon all |
| 12:50.10 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
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| 12:55.57 | laggo | how do i pattern match an extension with a name like 'sipgate-foobar'. the dialplan patterns wiki page says [a-z] and [A-Z] is supported, but i don't know if that means i can do regex-style [a-z\-] |
| 12:58.41 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:58.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:58.49 | *** part/#asterisk anebi (~anebi@91.207.191.17) |
| 13:00.29 | laggo | also, is it possible to do an infinite while loop with AEL? |
| 13:00.34 | verywiseman | when i invent party to conference by press ** , i disconnected , pls see line nu 32,33,34 in http://fpaste.org/sD2z/ |
| 13:00.59 | *** part/#asterisk Stratisphere (~Stratisph@unaffiliated/stratisphere) |
| 13:01.08 | verywiseman | sorry i follow instructions in http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 13:07.07 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
| 13:07.26 | PoTe | Good morning! |
| 13:07.49 | PoTe | I don't know how to approach a certain problem, would be grateful if I could get some help. |
| 13:08.48 | PoTe | I need to be able to launch an AGI script whenever a call is in a queue and is about to be picked up by an agent, key part is that I need to know which agent will pick up the call.. so I imagine the call must be already forwarded to his channel |
| 13:08.55 | PoTe | (this is in asterisk 1.4) |
| 13:09.12 | PoTe | now, I know you can pass an agi as an argument to the Queue command |
| 13:10.08 | PoTe | but I can find very little documentation on that and Im just doing blind testing at this point, when is the AGI launched? do I have access to channel variables? Can I know by then the recipient of the call? |
| 13:10.16 | PoTe | Sorry if this is a dumb question :) |
| 13:11.11 | PoTe | Any thoughts on this? |
| 13:20.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:29.41 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 13:33.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:36.08 | *** join/#asterisk garymc (~chatzilla@host81-139-156-105.in-addr.btopenworld.com) |
| 13:42.20 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:42.22 | verywiseman | when i follow instructions in http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO , problem appear when i press ** , where caller(2323) disconnected , and this is logs http://fpaste.org/sD2z/, where is problem? |
| 13:43.52 | skrusty | PoTe, does it need to be an AGI? What's it for? |
| 13:44.02 | skrusty | you could always use the manager to pickup the agent connection event |
| 13:50.06 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 13:58.20 | nicola_pav | hello. i run htop and i see there are many many of asterisk instances |
| 13:59.37 | binbash_ | that's correct |
| 14:00.08 | nicola_pav | trying to connect to asterisk CLI but it fails |
| 14:00.20 | nicola_pav | to solve this i need to kill asterisk and start it again |
| 14:00.26 | nicola_pav | this happens daily |
| 14:00.54 | nicola_pav | binbash_: the number of instances is huge |
| 14:01.03 | nicola_pav | any hints y this happen? |
| 14:01.30 | DeHackEd | are they multiple threads of the same process? this is quite normal |
| 14:02.30 | binbash_ | Depends, like DeHackEd told you, if it's the same process it's normal. |
| 14:02.37 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 14:02.45 | skrusty | lol :) |
| 14:02.54 | DeHackEd | Katty: have you tried fondling it? it likes that |
| 14:03.09 | binbash_ | lol |
| 14:03.15 | skrusty | source or package katty? |
| 14:03.26 | Katty | what is source??? |
| 14:03.30 | skrusty | ok... |
| 14:03.33 | DeHackEd | compiled from source |
| 14:03.35 | skrusty | install windows |
| 14:03.40 | skrusty | and forget about asterisk ;) |
| 14:03.43 | Katty | you guys are soooooo cute ^_^ |
| 14:03.46 | Katty | you actually think i'm serious |
| 14:03.53 | skrusty | bless |
| 14:03.54 | Katty | that is /adorable/ |
| 14:04.13 | binbash_ | ;D |
| 14:04.14 | Katty | GOOOD MORNING ASTERISK |
| 14:04.14 | skrusty | so... what's the issue then? |
| 14:04.30 | skrusty | dont say morning, you worried me... just had lunch :) |
| 14:04.37 | DeHackEd | insufficient love |
| 14:04.44 | Katty | please keep in mind the Asterisk Christmas Card Exchange cutoff time is December 10th for Interntional Mailing and December 15th for mail within the United States |
| 14:04.48 | binbash_ | Morning? it's 3:05 PM! |
| 14:04.52 | Katty | if you would like to join the christmas card exchange, please /query me for details! |
| 14:05.10 | russellb | Katty: how many people have signed up? |
| 14:05.24 | skrusty | drum-roll |
| 14:05.46 | Katty | russellb: not as many as i'd like. just 6. |
| 14:05.58 | Katty | russellb: however i've gotten a few people /on the side/ to give me their address. |
| 14:06.00 | russellb | well at least it's manageable :-) |
| 14:06.14 | Katty | russellb: if you want to send cards, you might contact people directly |
| 14:06.23 | russellb | nods |
| 14:06.47 | russellb | that sounds like work |
| 14:06.53 | Katty | not really. |
| 14:07.11 | russellb | :-p |
| 14:07.12 | Katty | just /query person OHAI WHAT IS YOUR ADDRESS FOR XMAS CARD PLZKTHX |
| 14:07.37 | Katty | i have 13 cards written up so far |
| 14:07.40 | Katty | with about 8 more to do today |
| 14:07.41 | nicola_pav | DeHackEd: asterisk gets stuck for some reason |
| 14:07.49 | russellb | Katty: I need to buy some cards, then! |
| 14:07.55 | nicola_pav | its not like that there are lots of instacnes and its is working |
| 14:08.02 | nicola_pav | it gets stuck |
| 14:08.05 | Katty | russellb: yes'r. recommend running to hallmark and buying a pack |
| 14:08.13 | russellb | yessss |
| 14:08.15 | nicola_pav | i check htop and i can see that there are so many instacnes |
| 14:08.24 | Katty | russellb: you can get them all the same and some packs have a 3 different types |
| 14:08.27 | nicola_pav | asterisk will not respond to ayhting |
| 14:08.38 | Katty | russellb: much better than 5 bucks a card. |
| 14:08.45 | russellb | Katty: quite. |
| 14:08.47 | Katty | russellb: target also has some adorable packs. |
| 14:08.50 | binbash_ | nicola_pav what is top telling you? |
| 14:09.03 | binbash_ | How is the server load |
| 14:09.12 | nicola_pav | lots lots of instacnes |
| 14:09.19 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.151.106) |
| 14:09.20 | nicola_pav | that's it |
| 14:09.20 | binbash_ | okay but.. how is the server load? |
| 14:09.22 | leifmadsen | dances |
| 14:09.53 | Katty | good morning leifmadsen |
| 14:10.19 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 14:10.39 | nicola_pav | binbash_ server load normal |
| 14:10.46 | binbash_ | Okay |
| 14:10.51 | nicola_pav | just stuck instances |
| 14:10.53 | binbash_ | And if you kill asterisk and start it again it owrks? |
| 14:10.57 | nicola_pav | i know that its normal |
| 14:11.03 | leifmadsen | Katty: ohai! |
| 14:11.10 | Katty | leifmadsen: how're you dear |
| 14:11.12 | nicola_pav | but the ones we get are really a lot |
| 14:11.21 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 14:11.22 | nicola_pav | yes |
| 14:11.23 | binbash_ | Ok but, if you kill & start it works? |
| 14:11.25 | binbash_ | Ok |
| 14:11.26 | nicola_pav | it starts to work again fine |
| 14:11.29 | binbash_ | Build from source or? |
| 14:11.35 | nicola_pav | but tomorrow it will happen again |
| 14:11.40 | nicola_pav | its happens for some time now |
| 14:11.50 | binbash_ | What version and how did you install it? |
| 14:11.59 | nicola_pav | from source |
| 14:12.05 | nicola_pav | its asterisk 1.2 |
| 14:12.08 | nicola_pav | its old i know |
| 14:12.12 | nicola_pav | but was working fine so far |
| 14:12.14 | binbash_ | Ah sorry |
| 14:12.18 | binbash_ | Don't have any experiance with 1.2 |
| 14:12.20 | nicola_pav | except for it crashing lately |
| 14:12.28 | binbash_ | Can't help you with that :-) |
| 14:13.30 | nicola_pav | :) |
| 14:14.01 | nicola_pav | this asterisk is running call center |
| 14:15.53 | cmnky | perhaps its time to consider an upgrade |
| 14:16.56 | nicola_pav | cmnky: i agree |
| 14:17.06 | nicola_pav | but it was working just fine |
| 14:17.13 | leifmadsen | Katty: oh not too shabby |
| 14:17.57 | nicola_pav | cmnky: if it was not working at all, upgrading would be the only option |
| 14:18.29 | cmnky | upgrading is the eventual option, no matter whats happening |
| 14:18.59 | nicola_pav | cmnky: thanks anyway |
| 14:19.22 | cmnky | yw anyway |
| 14:19.35 | justdave | wonders why there's a different tarball inside the SRPM than the one released on the website |
| 14:19.53 | justdave | I thought the point of an SRPM was to turn the actual released tarball into an RPM |
| 14:20.11 | leifmadsen | justdave: what versions are you seeing, and what are you expecting? |
| 14:20.25 | leifmadsen | justdave: all of that is done manually, so it is possible the version may not be updated for some reason |
| 14:20.31 | leifmadsen | points at Qwell |
| 14:20.41 | justdave | I expected to see asterisk-1.8.0.tar.gz |
| 14:20.50 | leifmadsen | and what did you see? |
| 14:20.50 | justdave | instead there's asterisk18-sources-1.8.0.tar.gz |
| 14:21.25 | justdave | which made it hard to just drop in the 1.8.1rc1 tarball, change the version number, and rebuild |
| 14:21.26 | leifmadsen | sounds like a script could probably be updated to match the naming method of the other script that gets run :) |
| 14:22.05 | leifmadsen | justdave: I'd suggest opening an issue then asking for that to be changed. There is a project in mantis for AsteriskNOW (which is also for RPM issues, etc...) |
| 14:22.19 | leifmadsen | it's just an oversite when two scripts were built by different people at different times |
| 14:22.43 | justdave | 1.8.1 final is out today sometime I'm guessing? |
| 14:23.04 | leifmadsen | justdave: yes, it is tagged, I just need to finish the release announcement and get it signed by a couple of devs |
| 14:23.09 | russellb | PDF ... taking ... forever .... |
| 14:23.14 | leifmadsen | russellb: +1 |
| 14:23.24 | leifmadsen | russellb: that's what happens when you write a huge book |
| 14:23.44 | leifmadsen | russellb: BOOK! |
| 14:23.44 | russellb | leifmadsen: 566 |
| 14:23.54 | leifmadsen | russellb: damn you and your auto-refresh script |
| 14:24.06 | leifmadsen | :) |
| 14:24.18 | leifmadsen | I currently have 19 new voicemails.... |
| 14:24.23 | leifmadsen | I should probably check those today |
| 14:24.55 | justdave | with a few minor changes to the spec file it looks like it builds from the real tarball, mostly. |
| 14:25.06 | Katty | HELLO DAVE |
| 14:25.07 | justdave | I'll post a patch to the spec when I file the bug |
| 14:25.16 | leifmadsen | justdave: that's even better! thanks! |
| 14:25.19 | Katty | tho you're not the dave i'm thinking of :< |
| 14:25.22 | justdave | did 1.8.1 remove the odbc stuff? |
| 14:25.26 | Katty | i miss the dave i'm thinking of. |
| 14:25.31 | fenrus | \o/ |
| 14:25.33 | leifmadsen | justdave: no... |
| 14:25.38 | Katty | HAI FENRUS |
| 14:25.44 | Katty | hugs fenrus |
| 14:25.45 | justdave | the spec file from 1.8.0 is looking for it and it didn't build... |
| 14:25.49 | justdave | maybe it's missing a build-dep |
| 14:25.51 | Katty | fenrus: you are on the christmas card exchange, yes? |
| 14:25.52 | leifmadsen | weird |
| 14:26.06 | leifmadsen | luckily i have nothing to do with the spec files or building RPMs, I just realize tarballs :) |
| 14:26.19 | justdave | that's probably one of those ones that the menuselect won't enable it if the prereqs aren't there and the spec forgot to specify the prereq |
| 14:26.39 | fenrus | Katty, well - i dont like papers.. :) |
| 14:26.43 | justdave | and the official build system probably has said prereq already |
| 14:27.01 | Katty | fenrus: so? |
| 14:27.07 | Katty | fenrus: it takes like 40 cents to mail a christmas card. |
| 14:27.11 | fenrus | Katty, a postcard is a paper :) |
| 14:27.19 | Katty | fenrus: and people are so overjoyed to get them |
| 14:27.21 | fenrus | Katty, would you like a postcard from me? :) |
| 14:27.26 | Katty | fenrus: i would :> |
| 14:27.30 | fenrus | s/postcard/christmascard |
| 14:27.35 | Katty | fenrus: and i'd love to send you a christmas card! |
| 14:27.52 | justdave | yeah, menuselect says ltdl is missing |
| 14:28.07 | leifmadsen | justdave: ya odbc definitely would need all the prereqs |
| 14:28.56 | justdave | adds libtool-ltdl-devel to the build prereqs |
| 14:29.38 | leifmadsen | weird that it wasn't already there, because that has always been required |
| 14:29.41 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 14:29.58 | justdave | probably already had it on the build system and just forgot to put it in the spec file |
| 14:30.05 | leifmadsen | perhaps yes |
| 14:30.34 | Lord_Rahl | ? for anyone how would I switch the callerid to display the name and not the number on incoming calls |
| 14:31.10 | leifmadsen | Lord_Rahl: you have to have that support from the phone company -- it'd just be automatic |
| 14:31.51 | Lord_Rahl | leifmadsen, Yes they are pushing both but on my ploycam 301 I only show the number |
| 14:32.23 | Lord_Rahl | in cli all I see pass is the number |
| 14:32.30 | leifmadsen | Lord_Rahl: look at the SIP trace then and see if the rpid and all that info is being sent. You may need "sendrpid=yes" enabled |
| 14:33.54 | Lord_Rahl | leifmadsen, that would be in sip.conf I can grep for it if need be |
| 14:33.57 | russellb | leifmadsen: we don't have connected line stuff anywhere in the outline ... :-/ |
| 14:35.09 | russellb | I might see about sneaking it in there somewhere |
| 14:37.42 | leifmadsen | russellb: oh snap... interesting. Ya we should really cover that as it's a huge deal. |
| 14:37.55 | leifmadsen | russellb: I should really sneak in some CURL() stuff for webservices into External Services too |
| 14:38.00 | russellb | heh |
| 14:38.20 | leifmadsen | I would love to have another month to work on the book... |
| 14:38.30 | leifmadsen | maybe I'll get lucky and have lots of time over christmas :S |
| 14:39.27 | leifmadsen | ok, task at hand -- release announcements and signing to get 1.8.1 released this morning |
| 14:39.41 | leifmadsen | second task at hand: triage 50+ issues |
| 14:40.19 | russellb | & |
| 14:47.17 | PoTe | skrusty: I actually need to interact with a web service, so yes, I think it needs to be an AGI. |
| 14:49.35 | justdave | ok, odbc solved with libtool-ltdl-devel, but it's complaining about freetds and speex also, and I have those installed already (-devel, too), so not sure why it's failing to detect them. |
| 14:50.00 | justdave | those were already in the BuildRequires |
| 14:50.04 | justdave | menuselect just isn't finding them |
| 14:52.02 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 14:52.36 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 14:58.31 | Lord_Rahl | [TK]D-Fender, I am having a hard time finding it in sip.cfg. would you know the function name ? |
| 15:01.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 15:06.21 | dmz | Lord_Rahl what are you trying to find? |
| 15:08.52 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:11.38 | eduzimrs | Where can i rack in chan_sip.c to change the statment of a peer when it unregister, i`d like the status "BUSY" |
| 15:13.00 | WIMPy | eduzimrs: Back to Plan A again? |
| 15:13.29 | eduzimrs | WIMPy uaheiuheai YEAP i gave up the another |
| 15:14.44 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:15.25 | *** join/#asterisk bkruse (~Brandon@75.76.105.124) |
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| 15:22.05 | eduzimrs | WIMPy u know where i change this value? |
| 15:22.26 | WIMPy | grep should be your friend |
| 15:23.09 | bkruse | waves |
| 15:25.17 | *** join/#asterisk secesh (~secesh@adsl-074-167-019-138.sip.sav.bellsouth.net) |
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| 15:42.10 | *** join/#asterisk matt_ (~secesh@adsl-074-167-019-138.sip.sav.bellsouth.net) |
| 15:43.22 | Katty | asterisk christmas card exchange cutoff date is in 3 days--sign up now!!! |
| 15:48.29 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:57.53 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 15:59.59 | anonymouz666 | If I have a fixed jitter buffer of 200ms and then I have 3 packets as follows: the first arrives in the jitter buffer (0) and then the second at 20ms and third has a great jitter and then arrives 190ms after... the third packet will be dropped |
| 16:00.08 | anonymouz666 | is this theory correct? |
| 16:02.56 | WIMPy | gould imaging that the lengt would matter. |
| 16:03.23 | WIMPy | If the packet has the usual 20ms, it would go up to 210ms. |
| 16:03.30 | WIMPy | urgs |
| 16:03.37 | WIMPy | could imagine that the length would matter. |
| 16:04.26 | anonymouz666 | so the third is dropped |
| 16:04.34 | justdave | using Dundi/ as a switch in 1.8.x seems to hang :| 1.8.1rc1 didn't seem to fix it. |
| 16:04.43 | justdave | commenting out the switch=> lines in the dialplan make the hang go away (but then obviously, calls that would have used extensions found that way don't go through) |
| 16:05.00 | justdave | sip reload no longer hangs though (it did in 1.8.0) |
| 16:05.13 | *** join/#asterisk af_ (~getsmart@78.134.21.216) |
| 16:05.52 | WIMPy | I'm using dundi all the time. And I'm pretty sure I've done a sip relaod as well sind going 1.8.0. |
| 16:06.04 | WIMPy | s/sind/since/ |
| 16:06.21 | justdave | yeah, it seems very strange that would be broken for real without anyone else complaining about it |
| 16:06.28 | justdave | wonders what else is f**ed up on his system |
| 16:07.05 | justdave | I'm installed from the RPMs though, I didn't build from source directly |
| 16:07.11 | justdave | WIMPy: did you build from source? |
| 16:07.21 | WIMPy | yes |
| 16:07.25 | Katty | dances |
| 16:08.17 | justdave | I guess the official RPMs are kinda new, maybe there's other stuff wrong with them than the tarball name that hasn't been caught yet. |
| 16:10.03 | leifmadsen | justdave: I don't think there have been any changes to pbx_dundi in a while |
| 16:10.21 | leifmadsen | I don't like using the dundi switch anyways... I much prefer the dialplan functions |
| 16:11.43 | justdave | I probably would, too. I bet those are new since this originally got set up (back in 1.2 I think) |
| 16:11.57 | justdave | the switch stuff always felt like kind of a hack to me for some reason :) |
| 16:12.42 | leifmadsen | justdave: ya I pretty much never (if ever) use switch |
| 16:12.50 | leifmadsen | it's an old way of doing things (1.2-ism really) |
| 16:13.35 | WIMPy | But ist short, easy and functional. I geuss doing the same in the dialplan requires quite a few lines. |
| 16:16.05 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:16.48 | justdave | doing it via dialplan I could check for aliveness of other servers and provide the user with a useful message like "this may be because the server hosting that extension is currently unreachable" instead of the generic "invalid extension" |
| 16:17.13 | justdave | more user-friendly :) |
| 16:19.04 | WIMPy | You would have to know the extensions in order to do so, which kind of defeats the purpose. |
| 16:19.27 | justdave | not necessarily. If all the servers are up, I can say "invalid extension" |
| 16:19.36 | justdave | if any of them are down I can tack on the "maybe because it's down" |
| 16:19.57 | *** join/#asterisk oneseventeen (~onesevent@web64.webfaction.com) |
| 16:20.35 | WIMPy | How would you know it could match on some dundi peer instead of really being invalid? |
| 16:23.49 | oneseventeen | setting up an asterisk box for the first time to test... getting failed calls in my softphone. Where should I start looking first? (I assume log files, but am not sure which one) |
| 16:25.01 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:25.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:27.46 | ChannelZ | Depends what a you mean by 'failed calls'. Look at the console first |
| 16:29.44 | oneseventeen | ChannelZ: thanks. Apparently I'm more noob than I thought and I forgot to set outbound routes... |
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| 16:32.01 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 16:35.52 | oneseventeen | ChannelZ: what is the console? |
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| 16:51.04 | justdave | isn't setting nat=yes on a sip.conf entry supposed to make it ignore the destination requested in the sip packets and send it back to the IP it got it from instead? |
| 16:51.53 | Corydon76-home | Generally, yes |
| 16:52.03 | dimm | who must be own of /dev/ttyUSB* ? |
| 16:53.15 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:55.33 | justdave | getting no audio on an external connection and rtp debug is showing it sending packets to an internal IP address |
| 16:55.41 | justdave | which isn't matched by our localnet= param |
| 16:55.57 | justdave | sip debug shows the IP it's sending to getting requested in the SIP INVITE |
| 16:56.24 | justdave | which asterisk *should* be ignoring and sending it back to the IP it got it from I thought, because of the nat=yes on that device entry |
| 16:57.14 | Corydon76-home | justdave: localnet=... is for an entirely different situation |
| 16:58.16 | Corydon76-home | localnet is for a situation where the machine does not have the actual external IP on the box, so it needs to know what IP to masquerade as when sending packets which are not to the localnet |
| 16:59.14 | Corydon76-home | nat=yes means the client to the server is behind a NAT... not that the server is behind a NAT |
| 16:59.22 | Lord_Rahl | [TK]D-Fender, Thanks I was able to get it to work :) |
| 17:01.06 | joeyjones | [TK]D-Fender: thjanks for last night, i've decided to just go with ulaw as i had tried to compoile before but had multiple failures |
| 17:01.33 | *** join/#asterisk imox1234 (~imox1234@p4FC5C4C7.dip0.t-ipconnect.de) |
| 17:02.26 | Corydon76-home | justdave: btw, after you install a new package, you must re-run ./configure |
| 17:02.49 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 17:02.59 | justdave | Corydon76-home: don't you have to re-run configure before you can compile it to begin with? |
| 17:03.14 | Corydon76-home | justdave: Run it for the first time, yes |
| 17:04.03 | Corydon76-home | But any time you change the dependencies on the box, configure is what finds (and caches) the location (or non-existence) of those dependencies |
| 17:04.23 | justdave | oh, yes, I did that. |
| 17:04.28 | justdave | blew away config.status first too |
| 17:04.49 | justdave | (so it wouldn't use the cached values from before) |
| 17:05.27 | Corydon76-home | If it's still not finding the package, then it's likely that either a) your packages are of insufficient versions, or b) your packages don't conform to our expectations (broken) |
| 17:06.01 | Corydon76-home | Configure won't use cached values from before |
| 17:06.20 | Corydon76-home | menuselect (and make) is what uses the cached values |
| 17:07.24 | justdave | ok, menuselect found ltdl after I installed it |
| 17:08.01 | justdave | it did not find freetds or speex (which were already installed before I started because they were listed in the BuildRequires in the spec file) |
| 17:08.32 | Corydon76-home | What version of Asterisk (and FreeTDS and SpeeX) are you using? |
| 17:08.54 | justdave | asterisk 1.8.1-rc1 |
| 17:09.10 | justdave | freetds-devel.x86_64 0.64-1.el5.rf installed |
| 17:09.22 | justdave | speex-devel.x86_64 1.0.5-4.el5_1.1 installed |
| 17:09.45 | Corydon76-home | You also need speexdsp |
| 17:10.36 | Corydon76-home | and I believe freetds now needs to be 0.82 or greater |
| 17:10.45 | justdave | that's listed as an alternative to speex on the menuselect page, and there's no package by that name available in rpmforge or RHN |
| 17:10.59 | justdave | that'd explain freetds though (version too old) |
| 17:11.17 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 17:11.21 | justdave | I did end up building it earlier without either of those (we're not using them anyway) |
| 17:11.49 | justdave | just passed the options to tell rpmbuild to ignore those subpackages |
| 17:11.52 | Corydon76-home | There are two functions that use speex, though |
| 17:12.13 | Corydon76-home | AGC() and DENOISE() |
| 17:13.37 | justdave | tries to remember the bot triggers |
| 17:13.38 | justdave | !nat |
| 17:13.43 | justdave | ~nat |
| 17:13.44 | infobot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 17:13.47 | justdave | there it is :) |
| 17:13.53 | justdave | oh, but "see docs" now. |
| 17:13.57 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:13.57 | justdave | that used to give a url to a web page |
| 17:14.36 | WIMPy | No, that was |
| 17:14.41 | WIMPy | ~sipnat |
| 17:14.41 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:15.20 | justdave | aha, thanks |
| 17:15.53 | *** join/#asterisk Tim_Toady (~moi@188.4.87.53.dsl.dyn.forthnet.gr) |
| 17:16.55 | justdave | I have three other servers this is working fine on, and I can't find anything nat-related different in the config |
| 17:17.03 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 17:17.19 | justdave | (all set up the same way, in different offices behind the same brand of firewall equipment in each place) |
| 17:17.41 | justdave | the only difference I can readily see is the working ones are all asterisk 1.4.x and the broken one is 1.8.1-rc1 |
| 17:18.16 | justdave | maybe a new config option in 1.8.x I'm missing or something |
| 17:22.22 | justdave | phone B in http://www.aocomputing.net/?p=3 is exactly the situation I have in all 4 locations. |
| 17:22.48 | justdave | and the suggested solution is how they're all set up, too |
| 17:23.25 | justdave | wonders why the 1.8 box is trying to send RTP traffic to the IP given by the client instead of the one the traffic actually came from |
| 17:23.44 | verywiseman | i want to exit from conference by press # , how can i do that? |
| 17:24.54 | justdave | verywiseman: "core show application meetme" from your cli |
| 17:24.58 | justdave | you probably want the "X" option |
| 17:25.18 | jdoe | justdave: reinvite? |
| 17:25.36 | justdave | canreinvite=no if that's what you're asking |
| 17:25.53 | WIMPy | or directmedia=no |
| 17:27.41 | justdave | ok, adding directmedia=no didn't have any effect on it |
| 17:30.00 | skrusty | Justdave: are you using realtime or static config files? and if realtime is caching enabled? have you done a prune? |
| 17:30.10 | justdave | it's static |
| 17:30.14 | skrusty | ok :) |
| 17:30.30 | jdoe | I'm not sure. I have nat peers on 1.8 with nat=yes, directmedia=no and qualify |
| 17:31.43 | brainiac | Does anyone know how I can get two extensions on a Polycom phone? |
| 17:32.41 | justdave | brainiac: sure, we do it all the time. as long as that phone has line buttons available |
| 17:34.01 | brainiac | justdave: it does (Polycom 330). I've been at it all morning and have yet to get it to work. |
| 17:35.26 | brainiac | I've played with the web interface and the xml config files; but when I call another ext., it calls from the same ext on both lines. |
| 17:35.32 | justdave | needs a series of reg.X entries in the phone config for each line |
| 17:35.42 | brainiac | did that |
| 17:35.49 | justdave | reg.1.XXXXX controls the first button, reg.2.XXXX controls the second, etc. |
| 17:36.24 | justdave | they're all params to the same <reg /> object in the xml, right? |
| 17:36.39 | brainiac | yeah |
| 17:36.40 | justdave | er, attributes I mean |
| 17:37.08 | justdave | you have a reg.X.lineKeys="X" on each one, too? |
| 17:37.40 | brainiac | reg.1.lineKeys="2" |
| 17:37.44 | brainiac | on both lise |
| 17:37.48 | brainiac | lines |
| 17:38.12 | brainiac | what does XXXXX mean (reg.1.XXXXX){ |
| 17:38.24 | justdave | I think that's telling it to use 2 buttons for each line |
| 17:38.29 | justdave | are there 4 buttons on the phone? |
| 17:38.44 | brainiac | 2 buttons |
| 17:38.45 | justdave | the XXXX was just a wildcard or whatever. there's several params starting with that |
| 17:38.48 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 17:38.55 | justdave | ok, you probably want lineKeys="1" on each one then |
| 17:39.03 | justdave | that's telling it how many buttons to use for that line |
| 17:39.09 | brainiac | ok |
| 17:39.13 | justdave | so if you only have 2 you just used them both on the first line with it set to t2 |
| 17:39.17 | justdave | -t |
| 17:39.57 | brainiac | I'm trying that nom |
| 17:39.59 | brainiac | w |
| 17:42.17 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:42.22 | brainiac | should I use a generic SIP setup for the other ext? |
| 17:45.16 | *** join/#asterisk jkroon (~jkroon@dsl-241-240-129.telkomadsl.co.za) |
| 17:46.31 | justdave | brainiac: needs the entire sip extension setup on both reg.X that you put in the phone. |
| 17:46.39 | justdave | as far as asterisk knows it'll be two separate phones |
| 17:47.03 | brainiac | ok |
| 18:12.46 | *** join/#asterisk dtoronto (~dtoronto@173-14-233-70-Utah.hfc.comcastbusiness.net) |
| 18:18.43 | verywiseman | i have meetme in macro, and i want if user exit from conference , go to extension inside macro, how? |
| 18:23.56 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 18:25.09 | brainiac | justdave: I got it working!! Thanks!! |
| 18:25.20 | justdave | brainiac: awesome |
| 18:28.33 | *** join/#asterisk saigop (3d0c11aa@gateway/web/freenode/ip.61.12.17.170) |
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| 18:33.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:34.29 | JerJer | has anyone attempted fax detection on a SIP channel? (using one DID for both voice and fax) (T.38 to asterisk fax for asterisk) |
| 18:35.42 | JerJer | faxdetect=incoming works great with who's your dahdi |
| 18:48.55 | bmoraca_work | has anyone here used Velocity Networks for wholesale VoIP? |
| 18:51.48 | justdave | ok, so trying to figure out how to do dundi lookups in dialplan instead of switch, and realized I basically have to have a catch-all extension that then does the dundi lookups... |
| 18:51.58 | *** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net) |
| 18:52.12 | justdave | and the presence of that extension will make things like IVR menus think any given extension exists |
| 18:52.24 | justdave | whether it does or not |
| 18:52.35 | bmoraca_work | you need to work on your contexts, then |
| 18:52.41 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 18:52.50 | *** join/#asterisk tmberg (tmberg@unaffiliated/tmberg) |
| 18:53.06 | justdave | well, if it's not in the same context, then the IVR menus would fail to detect extensions that are available via dundi |
| 18:53.46 | bmoraca_work | that's not true. do your dundi lookup on the dialed extension, if it doesn't exist, go to the i extension |
| 18:54.21 | justdave | aha. that's what I needed to know. Knew there had to be a trick to make that work. |
| 18:54.37 | *** part/#asterisk tmberg (tmberg@unaffiliated/tmberg) |
| 18:55.57 | justdave | and since I don't have a local i in my lookup context it'll fall back on the one defined by the ivr menu I assume. |
| 19:01.18 | verywiseman | i have meetme in macro, and i want if user exit from conference , go to extension inside macro, how? |
| 19:02.10 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
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| 19:27.51 | *** join/#asterisk flujan (~flujan@189.111.254.251) |
| 19:29.06 | flujan | hello guys. I am playing with asterisk 1.6.2.13 and I was reading the new manager manage_1.1 doc. |
| 19:29.17 | flujan | My asterisk box is not generating the event Transfer |
| 19:29.18 | flujan | http://pastie.org/1356524 |
| 19:29.36 | flujan | I am transfering using the built in transfer. |
| 19:29.41 | flujan | Not the sip transfer. |
| 19:29.53 | flujan | Is res_features suposed to generate the transfer? |
| 19:31.19 | p3nguin | You're wanting to use DTMF transfer instead of the transfer key on the phone? |
| 19:32.12 | flujan | p3nguin: yeap. |
| 19:32.31 | flujan | p3nguin: some guys here use a ATA so... |
| 19:32.31 | p3nguin | Did you add the t option to the Dial() command? |
| 19:32.50 | flujan | p3nguin: yeap. The transfer is working. The event on the AMI is not being displayed. |
| 19:32.52 | p3nguin | t and/or T, appropriately |
| 19:33.09 | p3nguin | Oh, I don't know anything about AMI events. |
| 19:43.57 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 19:46.04 | joeyjones | p3nguin: omfg, i upgraded to 1.6 |
| 19:46.49 | *** part/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 19:46.57 | p3nguin | Oh yeah? |
| 19:47.43 | joeyjones | p3nguin: apparently debian lenny backports has 1.6 |
| 19:48.09 | p3nguin | So what version did you end up with? |
| 19:48.14 | joeyjones | and it can convert to/from g722 |
| 19:48.27 | joeyjones | p3nguin: Asterisk 1.6.2.9-1~bpo50+3 built by buildd @ biber on a i686 running Linux on 2010-09-27 18:54:58 UTC |
| 19:48.29 | p3nguin | yeah |
| 19:48.51 | flujan | did you guys saw this event: http://pastie.org/1356524 |
| 19:51.54 | *** join/#asterisk kdas (43b41887@gateway/web/freenode/ip.67.180.24.135) |
| 19:52.34 | kdas | I am getting a "missing channel gtalk" error was i supposed to compile asterisk 1.8 with some special flag to enable it ? |
| 19:53.39 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 19:54.25 | p3nguin | kdas: Just make sure you go into the menu via make menuselect and enable chan_gtalk. |
| 19:54.55 | p3nguin | Usually if your system meets the requirements, it would be enabled by default. |
| 19:56.01 | kdas | p3nguin: i ahve both libncurses5 and ncurses5-dev install but menuselect/menuselect gives me "you need ncures installed" error |
| 19:56.45 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 19:57.06 | p3nguin | Try running make clean and then ./configure before running make menuselect. |
| 19:57.29 | kdas | p3nguin: and make menuselect reports "Terminal must be at least 80 x 27." and it is sure bigger than that |
| 19:58.11 | WIMPy | Maybe too big? |
| 19:59.11 | kdas | i got it |
| 19:59.11 | kdas | thanks |
| 20:01.43 | kdas | is there a way just to compile the chan_gtalk or do i have to recompile everything ? |
| 20:03.11 | p3nguin | If you did what I said, you'll end up compiling everything. If you had already met the requirements (which you clearly hadn't), and you simply forgot to enable chan_gtalk, then make should only make the chan_gtalk module. |
| 20:05.06 | yonahw | does anyone know what sip option 5 is? I have an issue where if a user is spying using chanspy and receives another call when that second call stops ringing the spy channel is dropped. Second call is not being answered. Phones are Polycom 550's. I see a notice immediately prior of NOTICE[27518]: chan_sip.c:4034 sip_setoption: Unknown option: 5 |
| 20:05.15 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 20:05.59 | kdas | cool thanks |
| 20:09.36 | exothermc | is there another codec that is similar to g.729 (in quality and bandwidth) but is open? |
| 20:10.07 | malcolmd | iLBC is generally regarded as the closest thing in OSS land |
| 20:10.09 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 20:10.16 | drfreeze | Anyone here have experience with polycom phones? |
| 20:10.23 | Naikrovek | drfreeze: yes |
| 20:10.26 | Naikrovek | lots of us |
| 20:10.49 | drfreeze | I'm trying to find out if the sidecar for the 650 supports BLF (ie, shows the state graphically or with colored LEDs) |
| 20:11.23 | Naikrovek | exothermc: what's so big about being "open"? do you drive an open source car or walk on open source pavement? |
| 20:11.31 | Naikrovek | drfreeze: yes it does |
| 20:11.40 | Naikrovek | i have a 650 and sidecar, and i use it exclusively for BLF |
| 20:12.01 | exothermc | Naikrovek: No I just don't want to pay for licensing. |
| 20:12.02 | drfreeze | Naikrovek: thx. does it have the color LED AND the graphic icon? |
| 20:12.08 | Naikrovek | drfreeze: yes |
| 20:12.10 | Naikrovek | exothermc: ah |
| 20:12.15 | drfreeze | cool, thanks |
| 20:13.04 | Naikrovek | exothermc: G722 uses same bandwidth as G711, has MUCH better sound. G729 has no open equivalents, but you could use speex or iLBC or maybe even gsm if you're really worried about bandwidth |
| 20:13.31 | brainiac | Can I put a TDM410P and an AEX410 in the same machine running an 8-trunk group? If so, do I have to use separate drivers? |
| 20:13.35 | WIMPy | G722 can use less bandwidth than g711 |
| 20:13.41 | Naikrovek | yes it can |
| 20:13.53 | Naikrovek | i think the max it uses is equivalent to g711 |
| 20:14.04 | WIMPy | yes |
| 20:14.06 | p3nguin | naikrovek: I'd imagine he cares more about being free rather than being open. |
| 20:14.11 | WIMPy | 48, 56 or 64 kbps |
| 20:14.27 | exothermc | Naikrovek: Ya 722 is all well and good, but only really matters if it carried A-Z. In my case I'm trying to solve a bandwidth bottleneck at an end point. |
| 20:14.41 | Naikrovek | exothermc: how many simultaneous calls |
| 20:14.49 | exothermc | Naikrovek: 92 |
| 20:14.59 | WIMPy | exothermc: G726? GSM? |
| 20:15.03 | Naikrovek | over what bandwidth? t1? |
| 20:15.23 | exothermc | Naikrovek: 2Mbit |
| 20:15.35 | Naikrovek | yeah wow |
| 20:15.37 | Naikrovek | okay |
| 20:15.38 | Naikrovek | E1 |
| 20:15.41 | Naikrovek | (or equiv) |
| 20:16.08 | Naikrovek | what does your provider support? or are you at both ends of that link |
| 20:16.15 | yonahw | has anyone seen sip_setoption unknown option 5 warnings with Polycom phones? Does anyone know what option 5 is supposed to be anyway? |
| 20:16.24 | Naikrovek | yonahw: w |
| 20:16.26 | Naikrovek | um |
| 20:16.27 | Naikrovek | no |
| 20:16.28 | Naikrovek | i mean |
| 20:16.53 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:17.17 | yonahw | Naikrovek: were you saying that you mean something else? never seen it? noone has ever seen it? |
| 20:17.30 | exothermc | Naikrovek: Well I'm at both "ends" so to speak, but this is going out a PRI at the far end so forced g.711 from the TDM. |
| 20:18.03 | Naikrovek | yonahw: never saw an option 5 before, i don't even thing sip options work that way |
| 20:18.12 | Naikrovek | the whole thing is not option 5 |
| 20:18.24 | Naikrovek | do a sip debug when it happens so you can see what asterisk is seeing |
| 20:18.40 | Naikrovek | that'll give you a better clue about this option (it isn't 5, i'm sure) |
| 20:19.30 | yonahw | Naikrovek: sounds like good advice. Started to do that before but nothing stuck out at me. Will try again and look more closely. |
| 20:19.41 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 20:19.43 | *** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:19.49 | Naikrovek | just debug the IP of the phone that you will test it with |
| 20:19.56 | Naikrovek | that'll cut down on the noise a LOT |
| 20:20.06 | Naikrovek | then pastebin the whole thing in here |
| 20:20.10 | Naikrovek | um |
| 20:20.19 | Naikrovek | pastebin the whole thing, and give us a link |
| 20:20.31 | Naikrovek | mention the line that you see the error on as well |
| 20:20.37 | Naikrovek | afk lunch |
| 20:23.35 | yonahw | Naikrovek: http://pastebin.com/u56AWLHR line 84 is the option notice |
| 20:23.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 20:34.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:37.21 | brainiac | Does anyone know if I can put a TDM410P and an AEX410 in the same machine running an 8-trunk group? |
| 20:41.05 | exothermc | brainiac: No reason you couldn't |
| 20:45.47 | *** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 20:49.25 | malcolmd | brainiac: yes, you can. |
| 20:56.50 | *** join/#asterisk mtltemplar (~sabayonus@75-146-143-126-albuquerque.hfc.comcastbusiness.net) |
| 20:57.04 | mtltemplar | hey question about extension dialing and playing a sound |
| 20:57.35 | mtltemplar | i have an asterisk system attached to a nortel PBX. that PBX has an extension (76) that when dialed opens a connection to my PA system and allows me to do whatever. |
| 20:57.53 | mtltemplar | I would like to be able to dial an extension on asterisk and have it dial that 76 then play an mp3 file and hang up |
| 20:57.56 | mtltemplar | is this possible? |
| 21:00.03 | p3nguin | yes |
| 21:11.23 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 21:12.15 | jsolares | i'd do it with a call file, or the AMI, generate a call into your nortel pbx extension 76 and connect that to a context/extension/priority in your asterisk that does what you want |
| 21:13.26 | jsolares | heck there's even an originate app now |
| 21:13.29 | mtltemplar | sorry. mostly new to the asterisk arena. so i can certainly dial 76 on the nortel but how do i connect it to a context (like Playback(soundfile.mp3))? |
| 21:13.34 | mtltemplar | originate? |
| 21:13.40 | mtltemplar | sorry for my ignorance |
| 21:14.03 | jsolares | Originate(Dahdi/g0/76,exten,mycontext,76,1) |
| 21:14.08 | p3nguin | Originate() "core show application Originate" |
| 21:14.43 | jsolares | then on [mycontext] and extension 76 you would do whatever you want |
| 21:15.21 | jsolares | i remember when you had to modify the asterisk source code itself to do it... XD |
| 21:15.46 | jsolares | i guess there's not much reason for generating callfiles inside an agi nowadays :o |
| 21:16.05 | p3nguin | Not really, unless you just like doing things the hard way. |
| 21:16.43 | jsolares | i wonder now since when is originate available |
| 21:18.01 | *** part/#asterisk clintc (~clintc@n128-227-48-55.xlate.ufl.edu) |
| 21:18.40 | p3nguin | since 2008-12-18 |
| 21:18.42 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 21:19.25 | jsolares | you sure? the voip-info wiki says 1.6.2 which is not that long after that |
| 21:19.41 | jsolares | heh i should go reread what all aplications do in 1.8 |
| 21:19.58 | p3nguin | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2-current |
| 21:20.10 | p3nguin | Scroll down to 2008-12-18 14:23 +0000 [r165433-165469] |
| 21:20.26 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:20.35 | jsolares | you don't have much to do, do you? :P |
| 21:21.07 | jsolares | most of our customers are very afraid to upgrade anything, if it works don't touch it, so 1.4.22 for them |
| 21:21.30 | jsolares | i'm really liking 1.8 |
| 21:21.58 | p3nguin | It didn't even take me 2 full minute to find that. |
| 21:24.11 | DeHackEd | is there a way (using any method... AGI, manager, etc) to detect if the calling system is a fax machine by the fax tones? I have a pretty basic SIP line coming in. |
| 21:28.44 | p3nguin | dehacked: Did you try the faxdetect setting? |
| 21:29.43 | DeHackEd | p3nguin: docs say only available through a ZAP channel. all I got is SIP |
| 21:30.05 | p3nguin | It's a sip.conf setting, so I doubt it's only on zap channels. |
| 21:31.22 | mtltemplar | hmm. no originate. im on 1.4.21 and not allowed to upgrade (manager...) |
| 21:31.40 | mtltemplar | so i should look at how to craft call files? |
| 21:31.43 | jsolares | mtltemplar, then you could use system() and have that generate a call file |
| 21:32.24 | p3nguin | If you're going to use System(), you might as well use originate inside it. |
| 21:32.40 | p3nguin | It'll be a LOT easier. |
| 21:32.47 | jsolares | he's using an older version that doesn't have originate |
| 21:32.53 | p3nguin | I doubt that. |
| 21:33.09 | p3nguin | It doesn't have app_originate. |
| 21:33.14 | jsolares | you said it yourself, 2008-12-18 in svn for what would become 1.6.2 |
| 21:33.17 | jsolares | he has 1.4.21 |
| 21:33.33 | p3nguin | originate was surely available in 1.4.21. |
| 21:33.52 | jsolares | nope |
| 21:33.58 | jsolares | just checked on a 1.4.21.1 |
| 21:34.09 | p3nguin | You're going to make me install it myself, aren't you? |
| 21:34.29 | jsolares | you're going to do it to yourself :P |
| 21:35.00 | mtltemplar | i did core show application originate, no such thing. did core show applications and it wasnt in the list |
| 21:35.17 | p3nguin | mtltemplar: Just type in originate and press enter. |
| 21:35.18 | malcolmd | DeHackEd: asterisk can detect CNG tone across a SIP channel, but I don't think there's any manager event output on that detection (i could certainly be wrong) |
| 21:36.20 | p3nguin | mtltemplar: What does that tell you? |
| 21:36.33 | p3nguin | The usage of originate, I presume. |
| 21:36.59 | jsolares | so it was in as a part of cli but not an app itself |
| 21:37.17 | p3nguin | (1533.08) <p3nguin> It doesn't have app_originate. |
| 21:37.19 | jsolares | yeah, doing a system (asterisk -rx 'originate blah blah') would be easier than learning the callfiles |
| 21:37.25 | p3nguin | quite |
| 21:37.25 | DeHackEd | malcolmd: I'm getting the hang of this, so long as asterisk does something predictable that's probably okay |
| 21:38.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:39.33 | malcolmd | jsolares: if you don't have app_originate.so in /usr/lib/asterisk/modules then you don't have the originate application. it's not just a part of the cli |
| 21:39.44 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:40.01 | malcolmd | DeHackEd: if it detects that it's a fax, it'll drop it into an extension in the current context called "fax" |
| 21:40.23 | jsolares | no app_originate.so, however i can issue an originate command in the cli |
| 21:43.29 | DeHackEd | malcolmd: I'll try it out later. now I have something to google for and experiment with |
| 21:43.54 | malcolmd | DeHackEd: good luck |
| 21:49.11 | brainiac | does anyone know how to test line voltage via a TDM-410P? |
| 21:59.22 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 21:59.31 | mtltemplar | yah that gives me originate's usages |
| 21:59.51 | p3nguin | mtltemplar: Put it inside a System() in your dialplan. |
| 22:00.47 | p3nguin | That's how we emulate app_originate on systems which did not have it. |
| 22:01.38 | mtltemplar | darn learning curve... |
| 22:02.04 | p3nguin | It's long, but not all that steep. |
| 22:02.32 | mtltemplar | ok. so first off i need to make an extension on my asterisk machine for 76 right? tell it to go to nortel/Zap/g2 or whatever right? then use a system(originate etc) in extensions.conf under an extension listing? |
| 22:03.04 | p3nguin | Recap what you need to happen. You dial ... and ... happens. |
| 22:03.33 | mtltemplar | say 9999. then i can dial 9999 and its configured activity is to send an mp3 to the nortel but originate the call from the configured 76 extension so it dials it first and plays it over my PA? |
| 22:04.58 | p3nguin | Let's assume the Nortel system is an IP PBX. exten => 9999,1,Dial(SIP/nortel/76) would allow you to call 9999 on Asterisk and it would dial to nortel's extension 76. |
| 22:05.24 | p3nguin | That's just for fundamental demonstration purposes. |
| 22:05.47 | p3nguin | How do you connect Asterisk to Nortel? |
| 22:07.22 | mtltemplar | through zaptel. its an old option11 |
| 22:07.42 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 22:08.17 | p3nguin | I'm not really familiar with how to dial it via zap, so maybe someone else will help out with that aspect... but if we continue the example using SIP I can try to give you a basis on how it will work. |
| 22:09.01 | mtltemplar | right so other extensions are configured as 9999,1,Dial(nortel/Zap/g2/{EXTEN:6}) |
| 22:09.27 | mtltemplar | take that back 5555555555,1,Dial(nortel/Zap/g2/{EXTEN:6}) |
| 22:09.44 | mtltemplar | not sure what the EXTEN:6 is but probably could just replace it with 76 as you were saying |
| 22:09.45 | p3nguin | exten => 9999,1,System(asterisk -rx "originate SIP/nortel/76 application Playback filename") |
| 22:10.24 | WIMPy | mtltemplar: Are you sure that nortel/ there is correct? |
| 22:11.19 | p3nguin | Using the System() command and originate, it will call extension 76 on nortel via SIP and run Playback(filename). |
| 22:11.58 | mtltemplar | this is the standard way an extension is listed in my current asterisk setup: exten => 5555555555,1,Dial(nortel/Zap/g2/{EXTEN:6}) |
| 22:12.32 | mtltemplar | p3nguin:sweet. and the file can be mp3? |
| 22:12.55 | mtltemplar | and do i need anything special to play the mp3 file like mpg123 or something on the system or does asterisk handle those file types natively? |
| 22:12.57 | p3nguin | {EXTEN:6} is invalid; it's ${EXTEN:6}. And ${EXTEN:6} would turn 5555555555 into 5555. |
| 22:13.30 | mtltemplar | sorry. you are right. trying to do this from memory as i walk back and forth between machines as i have no link to the asterisk server from my desk |
| 22:13.33 | p3nguin | If Playback() can correctly play back the mp3, it will work. I can't say if your system is capable of that or not. |
| 22:13.48 | mtltemplar | right, so just try it and see |
| 22:14.09 | p3nguin | If it fails, you could use different file formats until you can figure out how to make it play mp3s. |
| 22:14.19 | p3nguin | wav, gsm, sln |
| 22:14.32 | mtltemplar | right. i have the audio in mp3 and wav formats |
| 22:15.10 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 22:15.25 | *** join/#asterisk SirNeo (~sirneo@unaffiliated/sirneo) |
| 22:15.25 | p3nguin | I guess now we just need to figure out how to dial your Nortel via Zap. |
| 22:16.10 | p3nguin | It's probably something like Dial(Zap/g2/76), assuming that g2 contains channels which are connected to the Nortel box. |
| 22:16.16 | mtltemplar | right. i would assume i would just mirror another one already in there |
| 22:17.15 | mtltemplar | now to find an audio file that isnt my shelter in place audio so that people dont go scampering off to our conference rooms by accident... |
| 22:17.53 | p3nguin | Maybe they'd enjoy some music. |
| 22:20.25 | mtltemplar | you gonna be on for a bit p3nguin? |
| 22:20.49 | p3nguin | I'm usually in and out all day/evening, so probably. |
| 22:21.10 | mtltemplar | sweet. im gonna go give this a shot just wanted to know if i should wait until you were around... :) |
| 22:21.20 | mtltemplar | for when i screw it up. not if |
| 22:21.25 | p3nguin | haha |
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| 23:12.35 | adelas | can someone enlighten me about T.38 faxing? |
| 23:13.54 | adelas | I have Asterisk 1.4.36 connected to a T1 PRI line. A Fax machine connecting to analog-sip SPA2102 device that is T.38 faxing aware. |
| 23:14.14 | adelas | the asterisk version shoudl be able to do a faxing T.38 passthrough or something? |
| 23:14.19 | adelas | how does T.38 play in this? |
| 23:14.29 | adelas | when the machine sends a fax |
| 23:18.14 | *** join/#asterisk londonmet050 (~londonmet@unaffiliated/londonmet050) |