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| 00:05.50 | pmhaddad-lappy | is glad none of his pbxs are hosted |
| 00:06.21 | AJayMN | times like this i wish mine wasnt either.. |
| 00:08.04 | tfrew|afk | colo it in a datacenter near you that has 24x7 entrance |
| 00:09.18 | AJayMN | im in the middle of no where :P |
| 00:09.25 | AJayMN | 2 hours from Chicago |
| 00:09.58 | AJayMN | ok i dont think its zaptel.. zaptel is loading.. i dont have a app_meetme.so in modules or anything named meetme.. shouldnt i? |
| 00:10.47 | *** join/#asterisk moy (n=moy@CPE000d9d59afef-CM00140454d33e.cpe.net.cable.rogers.com) |
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| 00:12.58 | InHisName | leifmadsen: finished eatin' yet? |
| 00:13.30 | AJayMN | there is no meetme of anything in my "show modules" |
| 00:14.56 | KavanS | I've got 2 asterisk boxes linked....one is for a remote office, I need to setup dialing rules to utilize the primary asterisk box to dialout |
| 00:14.57 | drmessano | Yes, asterisk did not compile with Zaptel support |
| 00:15.12 | drmessano | So you can do 1 of 2 things |
| 00:15.13 | KavanS | is there anyway to setup a simple dial plan to forward any dial commands to the primary asterisk? |
| 00:15.22 | drmessano | Reinstall Asterisk, which you just did |
| 00:15.33 | drmessano | Or reinstall Zaptel, then asterisk |
| 00:16.12 | drmessano | or 3.. Speculate about it for days |
| 00:16.28 | AJayMN | I did both |
| 00:16.32 | AJayMN | nothing has changed |
| 00:16.35 | AJayMN | zaptel is showing its loading |
| 00:16.58 | drmessano | Which version of Zaptel? |
| 00:17.20 | AJayMN | 1.2.7? the latest 1.2 ver i could find off asterisk.org |
| 00:17.43 | drmessano | Did you do a ./configure on asterisk first? |
| 00:17.52 | AJayMN | 1.2.27 |
| 00:18.41 | AJayMN | far as i know there is no ./configure on asterisk 1.2 |
| 00:18.48 | AJayMN | just make clean; make install |
| 00:19.13 | AJayMN | or am i just screwed in the head? :P |
| 00:19.22 | drmessano | You need to run a configure |
| 00:20.37 | AJayMN | there is no ./configure in the asterisk install dir |
| 00:23.06 | AJayMN | nevermind... |
| 00:23.17 | AJayMN | did make clean;make install on asterisk again its working.. |
| 00:23.19 | AJayMN | mmm |
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| 00:46.38 | InHisName | I have been trying to use MixMonitor() to capture beginning with dialtones thru hangup of everything heard. It waits till answerd to start recording. |
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| 01:08.29 | kn0x | still no luck with my sangoma compile |
| 01:08.37 | kn0x | i tried making a deb |
| 01:09.23 | kn0x | wanpipe compile still fails @ make[2]: *** [/usr/src/wanpipe-3.3.16/kdrvtmp/sdla_tdmv.o] Error 1 |
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| 02:14.27 | kn0x | http://pastebin.ca/1410846 |
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| 03:38.54 | kn0x | [TK]D-Fender: any other ideas? |
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| 04:08.04 | KyleK | I have an FXO that rings 123@mybox when a call comes in, to make asterisk ring multiple phones do i want Dial(SIP/realphone&SIP/softphone,20) or callgroups or ring groups or some other grouping? |
| 04:09.19 | *** join/#asterisk ^GoKu-^ (n=aambriz@187.134.46.21) |
| 04:10.38 | jaytee | KyleK, the Dial method you posted will work but you could also use a ring group, suppose it's a matter of preference |
| 04:12.07 | jaytee | I use Dial to ring multiple phones. not sure if a pickup group rings all the phones by default, I think it just rings the destination phone but others in the pickup group can answer it. |
| 04:12.22 | *** join/#asterisk jacky_tw (n=jacky@220-136-126-57.dynamic.hinet.net) |
| 04:12.31 | jaytee | I should probably reread that part of the "manual" |
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| 04:15.53 | KyleK | http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 04:16.49 | KyleK | yea seems i want the long Dial function call |
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| 04:53.44 | [TK]D-Fender | kn0x: I still see you using the version that I did not tell you to try |
| 04:58.14 | jaytee | these young whippersnappers, they never listen to their elders |
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| 06:51.50 | xorl | Any of you have experience with Cisco 7960 IP Phones? |
| 06:51.57 | xorl | It just gets stuck Configuring IP |
| 06:52.11 | xorl | like DHCP is not pulling |
| 06:52.14 | xorl | It's really anoying |
| 06:52.49 | MaliutaLap | I have one |
| 06:53.18 | MaliutaLap | xorl: does it pull it's config from tftp? |
| 06:53.51 | xorl | yeah |
| 06:54.02 | xorl | well, actually, not 100% sure if it's hitting the TFTP server |
| 06:54.15 | MaliutaLap | xorl: what do the logs say? |
| 06:54.33 | MaliutaLap | if it's not hitting you've got a bigger problem that I've ever had |
| 06:54.35 | xorl | I don't see it hit, no |
| 06:54.47 | xorl | Probably the Cisco 871 blocking TFTP through nat |
| 06:54.49 | xorl | :( |
| 06:54.51 | MaliutaLap | do you have logs of it hitting the dhcp server? |
| 06:55.08 | MaliutaLap | you're doing all the conf through nat? |
| 06:55.17 | xorl | yeah, unfortunately. |
| 06:55.21 | xorl | Asterisk server is hosted remotely |
| 06:55.28 | xorl | Have not had time to bring it in house. |
| 06:56.00 | MaliutaLap | * doesn't matter, you should have your dhcp/tftp at lest inside ... the rest can be done via http |
| 06:56.12 | MaliutaLap | then how you deal with * is separate |
| 06:56.39 | MaliutaLap | should be easy to allow tftp through the firewall/nat though |
| 06:56.50 | MaliutaLap | if it's only one phone ... otherwise you're screwed |
| 06:57.25 | MaliutaLap | xorl: is it getting a dhcp lease? |
| 06:58.14 | MaliutaLap | xorl: once it is getting an IP and being pointed towards the tftp server it should be straight forward |
| 06:59.30 | xorl | hmm, no it's not even getting an IP address |
| 07:00.03 | MaliutaLap | is the DHCP server inside the NAT? |
| 07:00.33 | xorl | yes |
| 07:00.37 | MaliutaLap | and what firmware are you using? |
| 07:00.38 | xorl | The Cisco 871 is the DHCP server. |
| 07:00.50 | xorl | Not sure of the firmware on the phones, think it's the latest, let me double check |
| 07:01.08 | MaliutaLap | SIP? SCCP? MGCP? |
| 07:01.28 | MaliutaLap | they are all slightly different |
| 07:01.34 | xorl | SIP |
| 07:01.52 | xorl | P003-08-2-00 |
| 07:02.39 | MaliutaLap | yeah, close enough |
| 07:02.55 | MaliutaLap | have you looked at http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/3_1/english/administration/guide/7900set2.html#wp1022748 |
| 07:03.11 | MaliutaLap | you might need to tell the phone it should be using dhcp/bootp |
| 07:03.18 | xorl | ah |
| 07:03.49 | MaliutaLap | mine was doing it when I got it |
| 07:04.19 | MaliutaLap | the only issue I had after that was stuff missing on the tftp server and/or the tftp server missing |
| 07:04.49 | MaliutaLap | personally I think everything should use DHCP ... even if it's getting a static IP |
| 07:05.54 | MaliutaLap | you should go through the settings section of the phone menu before rebooting and pointing it a tftp server |
| 07:06.00 | xorl | hmm |
| 07:06.02 | xorl | says DHCP enabled |
| 07:06.05 | xorl | : Yes |
| 07:06.09 | MaliutaLap | and do you have the right files on the tftp server? |
| 07:06.09 | xorl | but under the DHCP Server |
| 07:06.12 | xorl | it says 0.0.0.0 |
| 07:06.17 | xorl | MaliutaLap: yeah, pretty positive of that |
| 07:06.29 | MaliutaLap | that's normal until it gets a lease |
| 07:06.39 | xorl | hmm, why isn't it getting a lease though |
| 07:06.45 | MaliutaLap | if it's not hitting dhcp it will go to it's last booted config |
| 07:06.56 | xorl | MaliutaLap: I nuked that config out |
| 07:07.14 | MaliutaLap | check the cable connection, that is possibly a show stopper |
| 07:07.25 | xorl | Port lights on on the switch end I know that much |
| 07:07.27 | MaliutaLap | if that port can't get to the DHCP server |
| 07:07.55 | drmessano | Are you using PoE? |
| 07:07.55 | MaliutaLap | and the switch is allowing it to hit DHCP? you can block DHCP with a firewall too |
| 07:08.31 | MaliutaLap | baby, we all use Edgar Allan ;P |
| 07:08.34 | xorl | drmessano: No, and MaliutaLap: DHCP is allowed through the switch |
| 07:08.47 | xorl | the desktops use the same settings, vlans are 100% identical |
| 07:08.53 | drmessano | xorl: Try the other ethernet port on the phone |
| 07:08.56 | MaliutaLap | vlans? |
| 07:09.01 | xorl | drmessano: the PC one or the SW one? |
| 07:09.05 | MaliutaLap | that could be your problem |
| 07:09.07 | drmessano | The SW one |
| 07:09.13 | xorl | drmessano: In the SW one atm |
| 07:09.23 | MaliutaLap | never the PC one |
| 07:09.44 | xorl | yeah, never have, otherwise i get Ethernet Disconnected |
| 07:09.47 | xorl | This is strange. |
| 07:09.52 | MaliutaLap | it seems to be more network than phone specific |
| 07:10.06 | MaliutaLap | what happens when you reset the phone? |
| 07:10.12 | xorl | Desktops can grab DHCP, and if I plug a PC into the switch port this vlan is connected to, it can pull DHCP as well |
| 07:10.20 | xorl | MaliutaLap: Just goes back to Configuring IP |
| 07:10.40 | MaliutaLap | and hangs there? |
| 07:10.48 | MaliutaLap | until timeout that is |
| 07:10.53 | xorl | it just keeps going |
| 07:11.01 | xorl | it went overnight without timing out |
| 07:11.35 | MaliutaLap | hmm, are you sure that cable is in properly? I have had issues with mine not making contact properly sometimes |
| 07:12.02 | xorl | yeah it's in |
| 07:12.13 | xorl | if it's not in i get "Ethernet Disconnected" |
| 07:13.45 | MaliutaLap | have you tried running netcat or something along the path to see if it's transmitting packets for DHCP? |
| 07:14.13 | xorl | No real way for me to, ports built into the wall, guys up the ceiling right into the switch then to the router from there. |
| 07:15.06 | MaliutaLap | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg43617.html |
| 07:15.13 | MaliutaLap | old but probably still valid |
| 07:15.32 | MaliutaLap | the VLAN might be your problem |
| 07:16.18 | MaliutaLap | it is possible put a bridge in before the wall and capture everything going over it, all you need is a *nix machine with 2 nics |
| 07:16.51 | drmessano | If the 871 is the DHCP server, it may very well be on VLAN1 and the phone on VLAN100 |
| 07:17.15 | drmessano | Wheres a dumber router would be much less aware |
| 07:17.18 | drmessano | Whereas |
| 07:17.32 | xorl | yeah, I had it on a dumber router before all this |
| 07:18.00 | drmessano | On the cisco, do a SH CDP NEIGHBORS and see if the phone is there |
| 07:18.14 | xorl | CDP is not enabled |
| 07:18.27 | MaliutaLap | xorl: set the VLAN manually |
| 07:18.55 | MaliutaLap | see if that gets you a result ... you'll also then need to ensure the VLAN is properly set in all the files on TFT |
| 07:18.58 | MaliutaLap | P |
| 07:20.13 | xorl | drmessano: I see them now |
| 07:20.23 | xorl | I enabled CDP and now see a few phones |
| 07:20.27 | drmessano | ok |
| 07:20.53 | MaliutaLap | xorl: reboot the phone and see what it does |
| 07:21.01 | xorl | see the phone now (it's mac showed up) |
| 07:21.15 | MaliutaLap | xorl: I seriously recommend bringing that tftp server inside the NAT |
| 07:21.28 | xorl | MaliutaLap: I will do that shortly. |
| 07:21.34 | xorl | Ordered a new 1U for inhouse use |
| 07:21.45 | drmessano | TFTP in general does not work thru NAT |
| 07:21.50 | xorl | Just moved offices, didn't have a rack before (ghetto office) I now have a fully fledged telco room |
| 07:21.52 | MaliutaLap | amen |
| 07:22.25 | xorl | SIP001319783780 Fas 0 161 H IP Phone Port 1 |
| 07:22.26 | MaliutaLap | xorl: in those circumstances I run minimal stuff like tftp on my desktop if I have to |
| 07:22.35 | MaliutaLap | since my machines don't reboot often |
| 07:22.39 | xorl | true |
| 07:22.43 | xorl | That is a good idea. |
| 07:22.46 | drmessano | *cough* pumpkin *cough* |
| 07:22.48 | xorl | I might do that, give me a second |
| 07:23.05 | MaliutaLap | yeah, it's sub-optimal and needs to be stopped ASAP |
| 07:23.13 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:23.34 | MaliutaLap | don't ever let anyone else know about it ... they come to expect "stuff" |
| 07:23.36 | xorl | yeah it's not getting a DHCP lease yet the system sees it |
| 07:23.54 | MaliutaLap | VLAN |
| 07:24.29 | MaliutaLap | sometimes VLANs can be a PITA |
| 07:24.29 | drmessano | You probably have the phones on VLAN 100 and no DHCP there |
| 07:24.46 | MaliutaLap | and it's not real security |
| 07:25.10 | xorl | should I just tell it vlan1 |
| 07:25.13 | xorl | and see if it grabs the lease? |
| 07:25.18 | MaliutaLap | yeah |
| 07:25.35 | MaliutaLap | you'll need to do _all_ the phones though :) |
| 07:27.09 | xorl | that's the admin vlan yeah? |
| 07:27.17 | MaliutaLap | yeah |
| 07:29.08 | xorl | hmmm |
| 07:29.22 | MaliutaLap | I blame the VLAN |
| 07:29.37 | xorl | wtf is with this network lol I think imma reset this router, it's lagging so bad |
| 07:30.27 | xorl | connection to the router is fast, but outside the network is lagggy |
| 07:31.22 | MaliutaLap | you intartubes are tied |
| 07:32.57 | xorl | Think it's the move from a valley -> downtonw |
| 07:33.00 | xorl | downtown* |
| 07:33.53 | MaliutaLap | wireless? |
| 07:34.56 | xorl | microwave yeah |
| 07:35.08 | MaliutaLap | good luck with that |
| 07:35.16 | xorl | it's been fine for the past year, now it's shite. |
| 07:35.34 | MaliutaLap | I know that the Brisbane CDB is a radio blackhole ... way past saturation point |
| 07:35.50 | xorl | hmmm |
| 07:36.04 | xorl | "Configuring VLAN" then goes to "Configuring IP" |
| 07:36.08 | xorl | and just sits |
| 07:36.16 | MaliutaLap | trees? kite fliers? buildings? you're screwed |
| 07:36.33 | MaliutaLap | put in a bridge and see what it's doing |
| 07:39.45 | xorl | ugh this phone is going to get thrown |
| 07:39.55 | xorl | I configured it to hit the local tftp server and it's not |
| 07:40.07 | xorl | I told it to hit VLAN1 (vlan id 1) |
| 07:41.29 | xorl | hmm |
| 07:41.29 | xorl | brb |
| 07:43.41 | xorl | drmessano: yeah, the phone mysteriously dissapeared from the cdp |
| 07:43.47 | xorl | now it's back n/m |
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| 08:01.55 | imcdona | is there a way to have asterisk register via an outbound proxy? |
| 08:02.22 | MaliutaLap | yeah, put the proxy in it's way |
| 08:02.32 | MaliutaLap | or just tell it the proxy is the host |
| 08:03.05 | imcdona | aside from that. I am trying to register with teltel and they require that you connect to a proxy and then the proxy registers to the domain |
| 08:03.34 | MaliutaLap | so tell * that the host is the proxy address |
| 08:04.23 | imcdona | I did that....but asterisk insists on putting: REGISTER sip:obproxy.teltel.com:9090 |
| 08:04.41 | imcdona | instead of REGISTER: sip:teltel.com |
| 08:04.59 | MaliutaLap | imcdona: show us the sip.conf |
| 08:05.03 | MaliutaLap | ~pb |
| 08:05.04 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 08:05.37 | xorl | MaliutaLap: yeah imma call it a night on this phone |
| 08:05.40 | imcdona | here is the setup: http://blog.voicebyip.com/?p=91 |
| 08:05.43 | xorl | it doesn't want to grab DHCP period |
| 08:05.57 | xorl | Wish it would just do it and get it over with, it's causing me such a fricken headache. |
| 08:06.48 | MaliutaLap | xorl: kill the VLAN |
| 08:07.04 | xorl | MaliutaLap: Why would I do that? It'll take down my current network for outgoing traffic haha |
| 08:07.31 | MaliutaLap | imcdona: that's not your sip.conf |
| 08:07.50 | MaliutaLap | imcdona: if I can't see your running config then I can't help you |
| 08:08.07 | MaliutaLap | imcdona: you can't expect help without information |
| 08:08.28 | MaliutaLap | xorl: or move the phones onto a non-VLAN segment :P |
| 08:08.48 | imcdona | http://pastebin.com/m77b02326 |
| 08:09.12 | imcdona | that post shows what the settings are. The register statement is what I just pasted |
| 08:09.36 | xorl | MaliutaLap: My router IS a vlan period lol |
| 08:09.45 | imcdona | it's tricky as well because the username has an @ symbol in it |
| 08:09.59 | MaliutaLap | imcdona: you're not helping your cause here |
| 08:10.04 | xorl | everything that goes through it, is on a vlan |
| 08:10.22 | imcdona | ?? |
| 08:10.36 | MaliutaLap | xorl: well I'm betting the VLAN is the prblem, you might need to configer another one for the phones |
| 08:10.53 | MaliutaLap | imcdona: that's not your whole sip.conf ... and the whole file will effect thigns |
| 08:11.10 | xorl | MaliutaLap: probably i'll look through it |
| 08:11.40 | MaliutaLap | imcdona: you might think you're being smart, but by not giving me what I need to see to help you are making me less and less likely to help |
| 08:11.41 | imcdona | I didn't think that registry statements referenced peers |
| 08:11.58 | MaliutaLap | imcdona: and I'm not the only one in here that feels this way |
| 08:12.11 | imcdona | sure...I will give it to you...I didn't realize it was relavent |
| 08:12.27 | MaliutaLap | wouldn't have asked for it if it wasn't |
| 08:13.46 | imcdona | http://pastebin.com/m5ae42353 |
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| 08:15.14 | MaliutaLap | and that's the whole file? no [general] section? |
| 08:16.19 | xorl | MaliutaLap: I give up on these phones. |
| 08:16.27 | imcdona | I am using freepbx so there are several sip confs. What in general would apply to this issue...srvlookup is off |
| 08:17.08 | imcdona | so when you specify a register string, you can specify the peer details to go with it as well? |
| 08:18.09 | MaliutaLap | right, freepbx not supported here |
| 08:18.14 | MaliutaLap | go elsewhere |
| 08:18.57 | imcdona | forget the user interface. it's asterisk. if you don't know the answer just say so. |
| 08:19.31 | xorl | MaliutaLap: i'm bypassing the switch now directly to the router, it's still not DHCP'ing ugh, imma just take a breather and check this in the morning, thank god for the weekend. |
| 08:26.30 | KyleK | imcdona: so you're trying to register as like us@er@domain.com? is that even workable for SIP? |
| 08:27.25 | imcdona | I think I found what I need to do....I am going to try the outboundproxy statement in the peer config. I was unaware that you could specify a peer config when registering |
| 08:27.30 | imcdona | http://www.voiptalk.org/products/asterisk-voiptalk-sip-trunk-registration-using-outbound-proxy-setup.html |
| 08:28.04 | imcdona | it is possile. I have it working in xlite |
| 08:30.31 | KyleK | asterisk does register => stuff instead of register=stuff also |
| 08:36.01 | jeffspeff | anybody here have any luck sending name and number CID with Teliax SIP ? I can only send number... |
| 08:36.34 | imcdona | jeff you can only send number. not name to the pstn |
| 08:37.34 | jeffspeff | imcdona, then what is the purpose of exten => _1NXXNXXXXXX/2100,1,Set(CALLERID(all)=Joe Smith <2095551212>) |
| 08:38.18 | imcdona | so when you call another non PSTN phone...like another extension. or another sip phone. |
| 08:39.00 | imcdona | even if you had a PRI connected to your * box it would still only take the number. There is a database that is maintained that maps number to names. You have to pay to have that mapping done |
| 08:39.46 | imcdona | http://www.voip-info.org/wiki/view/CallerID |
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| 10:28.08 | imcdona | IS it just me or is outboundproxyport broken? |
| 10:38.26 | imcdona | I think this may be the issue: http://bugs.digium.com/view.php?id=9665 |
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| 11:24.05 | l0st-soul | hello |
| 11:24.30 | l0st-soul | i come back with this problem i have ... i must be missing something important here |
| 11:24.40 | l0st-soul | debian lenny, default asterisk install |
| 11:25.02 | l0st-soul | i only configure 1 user in sip.conf, i login with ekiga successfully and dial voicemail at 8500 |
| 11:25.27 | l0st-soul | i change pin code, everything seems ok, excepted that voicemail.conf gets emptied after confirming new pin# |
| 11:25.51 | l0st-soul | before changing pin: -rw-r----- 1 asterisk asterisk 11723 2009-01-04 20:26 voicemail.conf |
| 11:26.06 | l0st-soul | after new pin confirmation: -rw-r----- 1 asterisk asterisk 0 2009-05-03 13:23 voicemail.conf |
| 11:28.46 | l0st-soul | of course, after restarting asterisk there are no voice mailboxes anymore |
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| 11:49.43 | l0st-soul | (i dont have the problem anymore if i compile latest asterisk by hand - debian package problem then) |
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| 13:28.05 | shyam_k | hi when i dial to the external pstn from a sip there is some noice but its perfect when i bypass an incoming call to sip and then talk.. |
| 13:28.48 | shyam_k | initially i thought its the problem of poor imitating hardware but it works fine if i am recieving the call.. |
| 13:28.55 | shyam_k | may be its a codec problem!;-) |
| 13:29.30 | shyam_k | how can i know the codecs involved in a call? |
| 13:29.40 | shyam_k | asterisk server comments doesnt seem to have them,. |
| 13:31.09 | shyam_k | okay the sipclient shows different codecs;-) |
| 13:31.18 | UQlev | shyam_k, "asterisk -vvvr" and watch |
| 13:36.39 | shyam_k | okay it was my sip clients' problem.. it was having gsm as default making it as g117u made it all clear! |
| 13:36.40 | shyam_k | telephony is such a great thing to play with .. can't sit tight to complete my project;-) |
| 13:37.57 | shyam_k | UQlev: yeah i have it vvv |
| 13:38.54 | shyam_k | now one another question.. how to get the id of the incoming call coming through dahdi? |
| 13:39.05 | shyam_k | vvv isn't showing the incoming number.. |
| 13:40.19 | UQlev | shyam_k, do you want variable? logfile contains IDs |
| 13:41.16 | shyam_k | okay.. |
| 13:41.36 | shyam_k | oh it requires a variable at dialplan.. okay.. now ill RTFM;-) |
| 13:42.05 | UQlev | yes, better read manual |
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| 15:04.30 | puzzled | hi |
| 15:05.15 | MaliutaLap | fi? |
| 15:06.43 | puzzled | if a somevar contains foobar how do I get only the foo part? it was something with ${somevar:2} but that deletes from the front. how do I make it delete from the end? |
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| 15:12.23 | putnopvut | puzzled: I think you can specify a negative value instead of a positive one. |
| 15:12.39 | putnopvut | ${somevar:2} or ${somevar:3} |
| 15:12.43 | putnopvut | oops |
| 15:12.51 | puzzled | putnopvut: heh got it. thanks |
| 15:12.54 | putnopvut | ${somevar:-2} or ${somevar:-3} |
| 15:12.57 | putnopvut | :) |
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| 15:57.34 | Kobaz | how do i get cdr's to post even if the call cannot be completed |
| 15:58.57 | Kobaz | CDR output unanswered calls: no |
| 15:59.04 | Kobaz | that's in the status, but where's the setting for it |
| 15:59.32 | Kobaz | er |
| 15:59.34 | Kobaz | i found it, never mind |
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| 16:17.27 | Kobaz | aughh |
| 16:17.44 | Kobaz | i can't get CDR(userfield) to actually set |
| 16:18.38 | Kobaz | i can set accountcode but not userfield |
| 16:22.18 | Kobaz | why would i only be able to set accountcode on a cdr? |
| 16:28.10 | puzzled | Kobaz: not sure if still needed but did you compile asterisk-addons with userfield support and have the proper database schema loaded that supports the userfield field? |
| 16:35.14 | [TK]D-Fender | puzzled: C'mon... you're asking him to follow directions now? |
| 16:36.04 | Kobaz | puzzled: userfield works fine, for certain calls |
| 16:42.55 | Kobaz | i'm using accountcode since i can't write to userfield |
| 16:43.07 | Kobaz | i have to bust out in a bit, i'll put together an example |
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| 18:13.07 | jeffspeff | i'm having trouble getting something to work right. i made a context that manual dials a certain number if you go to extension 220. The call comes into * fine, and calls the other number fine; but the other number has certain authentication prompts (user/pin) and it keeps having trouble with recieving the digits or something.... it works fine when i use the softphone on my computer, but when i call the exten from my cell phone |
| 18:13.07 | jeffspeff | <PROTECTED> |
| 18:15.29 | jeffspeff | [TK]D-Fender, any ideas? |
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| 18:17.56 | [TK]D-Fender | jeffspeff: Sound like you haven;t told what your call is coming IN on. |
| 18:18.22 | jeffspeff | [TK]D-Fender, what do you mean? |
| 18:18.51 | [TK]D-Fender | jeffspeff: Cell phones don't magically talk to *. They come in on some sord of INTERFACE. |
| 18:18.56 | [TK]D-Fender | jeffspeff: Details boy... |
| 18:19.00 | jeffspeff | lol |
| 18:19.02 | jeffspeff | sorry |
| 18:19.06 | jeffspeff | lets see. |
| 18:19.31 | jeffspeff | cell phone -> teliax -> sip -> * |
| 18:19.45 | jeffspeff | is that what you wanted to know? |
| 18:20.17 | [TK]D-Fender | jeffspeff: Go call from an analog phone through Teliax and test |
| 18:20.34 | [TK]D-Fender | jeffspeff: if that fails the firstlike suspect is having chosen the wrong dtmfmode. |
| 18:20.45 | [TK]D-Fender | jeffspeff: And yes, thats what I wanted to know |
| 18:21.30 | jeffspeff | [TK]D-Fender, don't have easy access to an analog phone... what value and where is my dtmfmode set? sip.conf? |
| 18:21.49 | [TK]D-Fender | jeffspeff: in your peer entry, and you'd set it to match what teliax |
| 18:21.52 | [TK]D-Fender | uses |
| 18:22.08 | jeffspeff | ok, dtmfmode=rfc2833 |
| 18:22.10 | [TK]D-Fender | jeffspeff: And yes, clear SIP SETTINGS take place in sip.conf |
| 18:27.26 | jeffspeff | [TK]D-Fender, just woke up... took me a second to access that part of brain. lol |
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| 18:27.47 | jeffspeff | [TK]D-Fender, what does teliax use for dtmf? |
| 18:28.57 | [TK]D-Fender | jeffspeff: https://teliax.com/pages/TheTeliaxTelegram |
| 18:29.07 | [TK]D-Fender | jeffspeff: You really could TRY to look you know |
| 18:29.42 | jeffspeff | [TK]D-Fender, i googled, and nothing it showed jumped out at me... i just asked in case you knew, while i was still searching... thanks |
| 18:30.10 | [TK]D-Fender | jeffspeff: I ggogled that in clearly less time that it took me to answer... |
| 18:30.25 | jeffspeff | lol |
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| 18:58.35 | KyleK | so the iax2 protocol is nat friendly? |
| 19:04.11 | [TK]D-Fender | KyleK: Slightly more than for SIP. |
| 19:04.33 | [TK]D-Fender | KyleK: Then again I've never seen you follow the directions you've been handed |
| 19:09.54 | jeffspeff | [TK]D-Fender, just curious... how would i get a context to enter digits on a call that it has made? something like exten=>s,1,dial(blah) exten=>s,n,wait(4) exten=>s,n,123 where 123 is the digits it's inputing into the dialed call after waiting 3 seconds |
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| 19:16.58 | [TK]D-Fender | jeffspeff: "core show application dial" |
| 19:17.25 | carrar | dial |
| 19:17.29 | carrar | that command sounds familure |
| 19:18.49 | carrar | though I like Irish Spring better |
| 19:21.29 | jeffspeff | [TK]D-Fender, this is what i'm trying to get working, but the later dials don't work http://www.pastebin.ca/1411431 |
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| 19:25.41 | carrar | jeffspeff |
| 19:25.45 | carrar | something like this you mean? |
| 19:25.46 | carrar | http://www.mail-archive.com/asterisk@uc.org/msg06551.html |
| 19:28.37 | jeffspeff | carrar, can i do D(wwwwww1234#wwwww432#) ? |
| 19:28.50 | [TK]D-Fender | jeffspeff: yes |
| 19:29.29 | carrar | jeffspeff, anything else I can google for you while I am here? |
| 19:30.11 | jeffspeff | lmao, no... out of curiosity... what did you google for that? |
| 19:30.17 | carrar | harley parts, backpacks, maybe lawn furnature? |
| 19:30.29 | carrar | "asterisk dial" |
| 19:30.49 | [TK]D-Fender | carrar: Not like I didn't point him to CLI for the answer already... |
| 19:31.10 | carrar | or maybe it was "asterisk dial send dtmf" |
| 19:31.27 | carrar | TK haha yeah, that requires reading more then 20 characters :) |
| 19:31.49 | jeffspeff | I saw the D in there, but i was looking at voip-info when he came back with the example site |
| 19:31.50 | carrar | Those letters are confusing |
| 19:32.04 | jeffspeff | carrar, hey, be nice |
| 19:32.16 | carrar | espically where it say's "Send the specified DTMF strings" in the show command |
| 19:32.22 | carrar | heh |
| 19:32.33 | carrar | ALL in good fun |
| 19:32.39 | carrar | at your expense of course |
| 19:32.58 | [TK]D-Fender | carrar: No, merely requires COPY&PASTE |
| 19:33.52 | [TK]D-Fender | jeffspeff: I'll be nice... FUCK THE FUCKING WIKI. You might want to thin twice about asking about the less than kind version of that :) |
| 19:34.01 | [TK]D-Fender | think* |
| 19:34.36 | carrar | I think it's cold leftover pizza time |
| 19:34.49 | KyleK | [TK]D-Fender: I've been following about 70% of instructions, I got a good handle on how the pieces I have work now |
| 19:35.40 | [TK]D-Fender | KyleK: 7)% following can still = %100 failure and you're not showing anything |
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| 20:00.58 | mmlj4 | one question... why not update the wiki if it's out of date? |
| 20:01.54 | mmlj4 | it's listed on http://www.asterisk.org/support, so apparently it's of some value |
| 20:02.49 | [TK]D-Fender | mmlj4: Go for it. I volunteer you :) |
| 20:02.59 | mmlj4 | no |
| 20:03.20 | mmlj4 | you continually complain, but that's as far as you're willing to go |
| 20:03.33 | mmlj4 | so as far as I'm concerned, your noise is just htat |
| 20:03.33 | mmlj4 | that |
| 20:04.33 | [TK]D-Fender | mmlj4: Just because that isn't where I'd send anyone first doesn't mean I have to fix it. Better docs are provided WITH asterisk. They already exist |
| 20:05.06 | [TK]D-Fender | mmlj4: Saying "product X = bad" doe not obligate anyone to fix it when product Y which is already out there is better. |
| 20:05.58 | mmlj4 | about the answer I'd expect from a vista user |
| 20:06.24 | [TK]D-Fender | mmlj4: Because? |
| 20:09.39 | jdblack | I like cheese |
| 20:10.01 | [TK]D-Fender | mmlj4: that WIKI is by no means official anyway. If I start my own and its incomplete does everyone hav to fix mine too? |
| 20:11.59 | [TK]D-Fender | looks back on the small mountain of broken code that KerryG posted on his blog back in the day |
| 20:12.13 | [TK]D-Fender | Maybe we should update his blog too. |
| 20:12.14 | jdblack | I am going to wear my canada hat today, in honor of [tk] |
| 20:13.00 | [TK]D-Fender | jdblack: then spell it "honour" like you're supposed to ;) |
| 20:13.16 | [TK]D-Fender | jdblack: And FFS capitalize the country's name! |
| 20:13.16 | jdblack | I would, but it's a vancouver hat. :P |
| 20:13.30 | jdblack | canada? That's not a country! That's USA subburbs! |
| 20:15.52 | [TK]D-Fender | brb |
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| 20:17.54 | [TK]D-Fender | jdblack: http://tinyurl.com/daw8dp |
| 20:18.03 | jdblack | uh oh |
| 20:18.22 | jdblack | heh. |
| 20:18.37 | jdblack | hard to top that on |
| 20:19.42 | jdblack | oh, except to say, "sure, but we're more evil. And in real life, the evil, heartless bastard wins" |
| 20:21.16 | jaytee | not to mention we have more land that's actually "livable" instead of frozen tundra and ice |
| 20:21.59 | jdblack | careful of what you say to them. They might cut down all their trees and throw toothpicks at us. |
| 20:22.32 | KyleK | is there a channel i where i can ask about toll free did providers? |
| 20:22.52 | jdblack | This is one of them, but google is better. |
| 20:23.16 | jdblack | prices change pretty fast. any information you get on irc is likely to be wrong. |
| 20:23.33 | KyleK | yea "toll free did canada" |
| 20:24.36 | KyleK | well I'm interested in service quality as well, like how well they are at fixing thier own problems etc |
| 20:24.37 | kn0x | [TK]D-Fender: figured out the wanpipe issue... it was the dahdi version |
| 20:24.51 | kn0x | had to use an older versoin |
| 20:25.53 | kn0x | now the drivers are all working great, but asterisk wont answer the zap channels |
| 20:25.57 | kn0x | err dahdi |
| 20:28.13 | kn0x | no output on the cli during ring |
| 20:28.41 | Kobaz | it seems that GoSub won't match patterns.... ie: GoSub(foo,1234,1) context foo { _XXXX => { ... } } |
| 20:28.42 | kn0x | wanpipemon says line is connected: Line : connected |
| 20:28.56 | Kobaz | i'll get GoSub sent to non-existant extension in context foo |
| 20:29.02 | Kobaz | even though 1234 will match |
| 20:29.18 | Kobaz | goto works fine, given the same arguments |
| 20:30.56 | kn0x | [TK]D-Fender: you around? |
| 20:45.58 | *** join/#asterisk [netman] (n=netman@51.Red-83-45-1.dynamicIP.rima-tde.net) |
| 20:50.39 | *** join/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 20:51.28 | andrewn | KyleK, I have a toll free DID with vitelity.net and it works well |
| 20:51.54 | ThatKidKel | if a call is set up, and asterisk removes itself from the media path, what's the easiest way to get re-invites sent to put it back in the middle (ie.. to transfer a call) |
| 20:52.46 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:53.31 | *** join/#asterisk tzafrir_laptop (i=tzafrir@89.221.8.55) |
| 20:57.16 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 20:58.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:04.37 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
| 21:10.48 | *** join/#asterisk kami- (n=user@unaffiliated/kami-) |
| 21:10.53 | kami- | hello |
| 21:11.06 | KyleK | ahoy |
| 21:11.21 | kami- | I have a hard time getting Asterisk to call out via AMI |
| 21:11.54 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 21:12.08 | KyleK | ThatKidKel: what kind of phone are you using? try transfering the call to an extension doing whatever the phone wants you to do |
| 21:12.22 | kami- | After login, I send an Action: Originate ... and always get failed |
| 21:22.40 | kami- | this is the AMI command and the log: http://paste.pchowtos.co.uk/?id=1731&key=6d4b0e0e09efa |
| 21:26.09 | KyleK | i dont know anything about AMI personally |
| 21:30.27 | kami- | KyleK: is there a better place to ask or is here the right place? |
| 21:32.00 | KyleK | well this is the right place for IRC support for *, could try a different method like a mailing list |
| 21:32.51 | kami- | KyleK: I remember having seen a Digium forum. Do you know how responsive they are? |
| 21:35.01 | KyleK | dunno never posted there |
| 21:35.17 | KyleK | so what works with that E1 bridge of yours, can you call in but not out? |
| 21:35.20 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 21:35.53 | kami- | <PROTECTED> |
| 21:35.55 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 21:36.41 | kami- | But when I use the AMI, when I send the same params as in the call file, the call returns with an error 'Originate failed' |
| 21:43.31 | *** part/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 21:45.01 | KyleK | AMI is some sort of web api? |
| 21:45.10 | Kobaz | no |
| 21:45.29 | kn0x | im getting chan_dahdi.c:8748 mkintf: Unable to get parameters: Inappropriate ioctl for device |
| 21:46.25 | kn0x | when i load chan_dahdi |
| 21:47.18 | Kobaz | kn0x: is the dahdi kernel driver loaded |
| 21:47.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:48.17 | kn0x | Kobaz: yes |
| 21:48.44 | kn0x | dahdi, wanpipe, and ahdi_echocan_mg2 |
| 21:49.24 | kami- | KyleK: AMI is the Asterisk Manager Interface. A socket based protocol for sending commands to * |
| 21:50.11 | Kobaz | kn0x: does asterisk have permissions to the device... did you run dahdi_cfg |
| 21:50.32 | *** join/#asterisk ifnotwhynot (n=davidh@196-209-194-186-rrba-esr-3.dynamic.isadsl.co.za) |
| 21:50.44 | Kobaz | ifnotwhynot: and since when |
| 21:50.57 | ifnotwhynot | what do u mean? |
| 21:51.00 | Kobaz | heh |
| 21:51.04 | Kobaz | it's quote from a movie |
| 21:51.05 | ifnotwhynot | forever lol |
| 21:51.21 | Kobaz | hudsucker proxy |
| 21:51.21 | ifnotwhynot | are u a * boffin? |
| 21:51.28 | Kobaz | if not, why not! and since when?! |
| 21:52.01 | Kobaz | i like muffins |
| 21:52.04 | kn0x | Kobaz: i ran dahdi_cfg.. no output. |
| 21:52.13 | Kobaz | yeah |
| 21:52.19 | Kobaz | it wont have much output unless you do dahdi_cfg -vv |
| 21:52.37 | kn0x | Kobaz: I chown'd /dev/dahdi for asterisk:asterisk |
| 21:52.58 | ifnotwhynot | where can one set the time * waits for digits on outgoing calls |
| 21:52.59 | Kobaz | what dahdi version and what asterisk version |
| 21:53.14 | kn0x | sec |
| 21:53.19 | kn0x | asterisk 1.6.1.0 |
| 21:53.41 | kn0x | dahdi-linux-2.1.0 |
| 21:54.08 | Kobaz | mmm |
| 21:55.04 | *** join/#asterisk Ave_Rapina (n=user@87.196.33.47) |
| 21:56.25 | Kobaz | ifnotwhynot: for zap/dahdi or what |
| 21:56.59 | kami- | does somebody know what this means: 'Channel 0/1, span 1 got hangup request, cause 31'? |
| 21:57.05 | ifnotwhynot | i have * between the telco and legasy pbx, outgoing calls from legasy if the guy takes to long to complete the dial of the number it fails to make a call |
| 21:57.13 | ifnotwhynot | zap pri |
| 21:59.14 | Kobaz | ifnotwhynot: there's a digittimeout thinger in there somewhere |
| 21:59.19 | Kobaz | look at the chan_dahdi example |
| 21:59.28 | Kobaz | legacy |
| 21:59.36 | kn0x | Kobaz: is this a version problem? |
| 22:00.52 | Kobaz | kn0x: no idea |
| 22:01.08 | Kobaz | kn0x: try dropping down to 1.6.0.x or bumping up to a 1.6.2 beta |
| 22:01.17 | Kobaz | and try a different dahdi driver |
| 22:01.24 | Kobaz | like 2.1.0.4 is out |
| 22:01.36 | Kobaz | or maybe even newer, i havent checked |
| 22:02.38 | kn0x | Kobaz: the release canidates dont work with wanpipe driver |
| 22:04.18 | Kobaz | the last asterisk i got working with wanpipe was 1.4.22 |
| 22:04.38 | kn0x | Kobaz: anything I'm missing in 1.6? |
| 22:04.45 | kn0x | no idea what the difference is |
| 22:05.00 | kn0x | im new to the tdm shit. |
| 22:05.21 | kn0x | most of my servers still run 1.2 |
| 22:05.23 | kn0x | haha |
| 22:05.25 | Kobaz | heh |
| 22:06.16 | Kobaz | basically you need |
| 22:06.20 | Kobaz | libpri, dahdi, asterisk |
| 22:06.25 | Kobaz | and wanpipe |
| 22:06.31 | Kobaz | the problem is, wanpipe for dahdi is beta |
| 22:06.43 | Kobaz | so if you want to use this stuff in production, i highly recommend 1.4 |
| 22:06.49 | Kobaz | and sticking with zaptel |
| 22:06.55 | Kobaz | which means 1.4.21.2 is the latest you can use |
| 22:07.01 | kn0x | oh. |
| 22:07.16 | kn0x | shit yeah i didnt know i could still use zaptel |
| 22:07.17 | Kobaz | 1.6 has been running great for me, with pure voip |
| 22:07.23 | kn0x | this dahdi name is driving me nuts |
| 22:07.27 | Kobaz | but i haven't tried anything with dahdi |
| 22:07.37 | Kobaz | hah yeah, it's kinda dumb |
| 22:07.48 | kn0x | gergh |
| 22:07.49 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 22:07.53 | shmaltz | hi everyone |
| 22:07.55 | kn0x | so i have to start all the way over. |
| 22:07.56 | Kobaz | zaptel is only supported as far as 1.4.21.2 |
| 22:08.02 | Kobaz | it's pretty quick |
| 22:08.12 | kn0x | it took me two days to get dahdi/wanpipe happy |
| 22:08.13 | kn0x | :| |
| 22:08.15 | Kobaz | aww |
| 22:08.20 | Kobaz | yeah that's the problem |
| 22:08.20 | kn0x | haha |
| 22:08.21 | shmaltz | now that nufone is gone, anyone know a good provider pay as you go for $0.02 a minute US and doesn't limit my channels? |
| 22:08.29 | Kobaz | shmaltz: voip.ms |
| 22:08.30 | kn0x | shmaltz: vitelity... |
| 22:08.35 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 22:08.43 | shmaltz | kn0x, how long you been using them? |
| 22:08.55 | kn0x | id say more than a few years |
| 22:09.06 | shmaltz | kn0x, where they located? |
| 22:09.10 | kn0x | i started when they were sixtel/iax.cc |
| 22:09.27 | kn0x | shmaltz: their hq is in co, but i dunno about their servers |
| 22:09.33 | Kobaz | shmaltz: about a cent a minute, unlimited channels |
| 22:09.41 | shmaltz | kn0x, any commitments I have to make? |
| 22:09.48 | kn0x | Kobaz: voip.ms huh? |
| 22:09.51 | Kobaz | yeah |
| 22:10.04 | Kobaz | i've been playing with them for about a month now, they seem pretty good |
| 22:10.20 | kn0x | shmaltz: $40 to open an ccount i thinkl |
| 22:10.22 | shmaltz | stuck at the continue screen |
| 22:10.30 | kn0x | shmaltz: i just started playing with flowroute |
| 22:10.35 | Kobaz | shmaltz: voicepulse has 4 channels with every did, most destinations are less then a cent |
| 22:10.40 | shmaltz | kn0x, and maintaining each month, any commitments requreed |
| 22:10.51 | kn0x | shmaltz: no. just a prepaid balance required. |
| 22:10.54 | shmaltz | kobaz, too cheap to rely they'll be around for long |
| 22:11.08 | Kobaz | they've been around for a while |
| 22:11.12 | Kobaz | i've been using them for about a year |
| 22:11.20 | Kobaz | been down once |
| 22:11.21 | shmaltz | wow: |
| 22:11.23 | shmaltz | I'm sorry, but due to recent problems with online transactions, we are no longer accepting any customers from our website. If you are interesting in opening an account with Vitelity, please call us at 1-888-898-4835. We will ask that you fax us personal identifiable information. |
| 22:11.43 | shmaltz | kobaz, you obviously didn't understand what I said |
| 22:11.46 | kn0x | shmaltz: wow. never seen that |
| 22:12.08 | Kobaz | shmaltz: if you wrote complete sentences, maybe i would have :P |
| 22:12.19 | kn0x | shmaltz: asterlink is pretty popular. i have a client with some *big* traffic on them |
| 22:12.28 | shmaltz | shmaltz>kobaz, too cheap to rely they'll be around for long |
| 22:12.47 | Kobaz | yeah, that statement is missing some words |
| 22:12.52 | shmaltz | kn0x, they allow chaning CID on demand? |
| 22:13.03 | Kobaz | shmaltz: i still have no idea what you mean |
| 22:13.13 | kn0x | shmaltz: vitelity, flowroute, and asterlink do |
| 22:13.17 | shmaltz | kobaz, it's simple english |
| 22:13.29 | Kobaz | it's missing a verb or noun in there somewhere |
| 22:13.37 | kn0x | Kobaz: what version of zaptel should i use? |
| 22:13.39 | Kobaz | kn0x: do you know what he means? |
| 22:13.43 | shmaltz | kobaz, if one can't make a living out of the money I/we pay for the service they provide then they will NOT be around for long |
| 22:13.45 | Kobaz | kn0x: latest |
| 22:14.07 | kn0x | Kobaz: umm, yeah. i think he means that ultra-cheap providers dont allways last very lone |
| 22:14.21 | Kobaz | kn0x: yeah. now i get it |
| 22:14.46 | kn0x | Kobaz: any idea how to cleany deinstall dahdi |
| 22:14.49 | kn0x | no make uninstall |
| 22:14.57 | Kobaz | kn0x: rmmod the modules |
| 22:15.08 | kn0x | oh shit i see |
| 22:15.10 | kn0x | nvm |
| 22:15.20 | Kobaz | dont really need to uninstall it |
| 22:15.26 | Kobaz | eventually you'll move to dahdi anyway |
| 22:15.38 | kami- | where can I lookup the 'cause' in '-- Channel 0/1, span 1 got hangup request, cause 31' |
| 22:15.41 | kn0x | Kobaz: these are going to be production machines... |
| 22:15.53 | kn0x | they will be imaged and wont be updated for years |
| 22:15.58 | Kobaz | you can remove the dahdi_* stuff from /usr/sbin |
| 22:16.15 | xheliox | kami-: Google for isdn pri cause codes |
| 22:16.22 | Kobaz | and the kernel modules from /lib/modules/`uname -r`/ |
| 22:16.40 | kami- | xheliox: thank you |
| 22:16.58 | Kobaz | shmaltz: i doubt they are problems paying their bills, they obviously make enough money to sustain, since they've been around for quite a bit |
| 22:17.32 | shmaltz | Kobaz, or they have hopped to enough upstream providers that it didn't YET catch up with them. |
| 22:17.54 | Kobaz | s/are/have |
| 22:18.05 | Kobaz | possibly |
| 22:19.47 | Kobaz | or they just get good rates |
| 22:20.18 | kn0x | Kobaz: so 1.4.30 is no good with zaptel/ |
| 22:20.27 | Kobaz | buddy of mine runs a clec in florida, their costs are about .1 cents a minute |
| 22:20.32 | Kobaz | kn0x: no zaptel support |
| 22:20.36 | Kobaz | it was removed in 1.4.22 |
| 22:21.00 | Kobaz | shmaltz: but they push multi million amounts of minutes |
| 22:21.41 | kn0x | Kobaz: wait so 1.4.22 is no good? |
| 22:21.41 | shmaltz | kobaz, then it might be quicker than i thought till they go belly up |
| 22:21.47 | Kobaz | no zaptel in 1.4.22 |
| 22:22.36 | kn0x | oh so i should go to 1.4.21.1? |
| 22:22.42 | Kobaz | shmaltz: why would they go belly up... buying minutes at .001 dollars, and selling them at .005 dollars, = profit... who knows how many minutes they push |
| 22:23.27 | shmaltz | kobaz, show anyone paying .001 per minute to ANY LATA in the US, I think recip comps are more than that for those who can collect it |
| 22:24.23 | shmaltz | A French doctor says 'Medicine in my country is so advanced that we can take a kidney out of one man, put it in another, and have him looking for work in six weeks.' |
| 22:24.31 | shmaltz | A German doctor says 'That is nothing; we can take a lung out of one person, put it in another, and have him looking for work in four weeks. |
| 22:24.42 | shmaltz | The Russian doctor says 'In my country, medicine is so advanced that we can take half a heart out of one person, put it in another, and have them both looking for work in two weeks. |
| 22:24.52 | shmaltz | An American doctor, not to be outdone, says 'You guys are way behind. We recently took a man with no brains out of Illinois, put him in the White House, and now half the country is looking for work.' |
| 22:25.31 | kn0x | shmaltz: -_- |
| 22:26.01 | shmaltz | ok, I signed up for asterlink, waiting for their email for a while now |
| 22:26.40 | Kobaz | i forgot the website, but, they have some damn good rates |
| 22:27.42 | KyleK | so voip termination is outgoing calls? |
| 22:28.07 | Kobaz | depends on the point of view |
| 22:28.26 | KyleK | <PROTECTED> |
| 22:28.30 | KyleK | I love that |
| 22:28.36 | Kobaz | heh |
| 22:28.46 | Kobaz | human voice signals |
| 22:28.48 | KyleK | VoIP technology converts puny earth human wailing to digital packets |
| 22:29.31 | kami- | just for the records: I solved my problem by removing the CallerID packet from the Originate command |
| 22:30.15 | kami- | seems like the line doesn't support setting the caller id |
| 22:30.22 | kami- | thank you |
| 22:30.24 | kami- | good night |
| 22:31.45 | kn0x | st for the records: I solved my problem by removing the Callest for the records: I solved my problem by removing the Calle |
| 22:31.48 | kn0x | oops |
| 22:32.00 | KyleK | clipboard ftw |
| 22:33.52 | KyleK | asterlink's site could use a few more details |
| 22:33.56 | kn0x | Kobaz: |
| 22:34.01 | kn0x | you broke my setup -_- |
| 22:34.03 | kn0x | FATAL: Error inserting wanpipe (/lib/modules/2.6.26-2-amd64/kernel/drivers/net/wan/wanpipe.ko): No such device |
| 22:34.32 | KyleK | is that the wanpipe.ko from ZapTel? |
| 22:35.30 | Kobaz | i broke it? |
| 22:36.03 | Kobaz | kn0x: you'll need the following: compile zaptel, recompile wanrouter for zaptel |
| 22:36.41 | Kobaz | and before doing that unload zaptel/wanrouter |
| 22:37.47 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 22:38.59 | kn0x | Kobaz: that wat i did |
| 22:39.03 | kn0x | i compiled zaptel first |
| 22:39.36 | *** join/#asterisk [netman] (n=netman@204.Red-88-25-138.staticIP.rima-tde.net) |
| 22:40.28 | Kobaz | k |
| 22:40.33 | Kobaz | did wanrouter build nicely |
| 22:40.49 | kn0x | this came up during ./Setup |
| 22:40.54 | kn0x | after the compile i think |
| 22:42.24 | kn0x | yeah it Compiles Wanpipe Driver and the utilities |
| 22:42.39 | Kobaz | and then you do /etc/init.d/wanrouter start |
| 22:43.02 | kc8pxy | is odbc the only db "driver" for asterisk? or are there more specific bindings available? |
| 22:43.29 | kn0x | doh! |
| 22:43.33 | kn0x | zaptel modules didnt load |
| 22:43.54 | Kobaz | whats the error |
| 22:44.01 | kami- | arrgghh. I have a remaining problem. |
| 22:44.15 | kami- | the quality of the voice is very bad. |
| 22:44.44 | kami- | I can remember that there was a way to see which audio codec/conversion is used. |
| 22:45.27 | Kobaz | codecs= |
| 22:45.33 | Kobaz | in sip/iax configs |
| 22:45.41 | Kobaz | er |
| 22:45.47 | Kobaz | no wait |
| 22:45.51 | Kobaz | allow= and disallow= |
| 22:46.19 | kami- | Kobaz: and for pri? |
| 22:47.02 | KyleK | shmaltz: when did you get that "no new customers from website" error from vitelity? its giving me a page for signing up |
| 22:47.25 | Kobaz | i think codec is always going to be ulaw on a pri |
| 22:47.31 | shmaltz | KyleK, after hitting continue their tos |
| 22:47.57 | kn0x | Kobaz: |
| 22:48.16 | kn0x | codec_zap wont compile |
| 22:48.47 | kn0x | oh |
| 22:48.55 | kn0x | oh thats for the codec card |
| 22:48.58 | kc8pxy | kami-: where is the voice very bad? |
| 22:49.20 | kami- | kc8pxy: the wav file which is being played |
| 22:49.38 | kc8pxy | kami-: wimpy transcoder? |
| 22:50.40 | kami- | kc8pxy: I don't understand. |
| 22:51.11 | ifnotwhynot | where do you set clock on Pri? |
| 22:51.33 | ifnotwhynot | or show the available clocks |
| 22:51.56 | Kobaz | ifnotwhynot: straight zaptel driver, or wanpipe? |
| 22:52.27 | ifnotwhynot | is there a command to check for available clocks? |
| 22:53.23 | kc8pxy | kami-: you are saying that someone dialing in, is played a wav file? |
| 22:53.54 | kami- | kc8pxy: yes Playback(file-name) |
| 22:54.17 | kc8pxy | kami-: what codec is being sent out of the pri? |
| 22:54.55 | kami- | kc8pxy: I didn't configure anything knowingly. Can I see in the logs which codec is being used? |
| 22:55.55 | kn0x | update-rc.d: warning: /etc/init.d/asterisk missing LSB information |
| 22:57.28 | Kobaz | ifnotwhynot: you cant view the clock info, you just set it to what it's supposed to be, depending what your other side is set to |
| 22:57.54 | kc8pxy | kn0x: "warning! Warning! I can't tell what distro you're using!" is what your asterisk is saying with that. |
| 22:58.33 | kc8pxy | kn0x: unles LSB refers to Lower sideband. |
| 23:01.38 | kc8pxy | kami-: what i meant by a wimpy transcoder, is unless your wav file is being sent out the pri in raw wave format, your asterisk server has to translate it into the codec that IS headingout the pri, and if it doesn't have enough horsepower, that can kill yoru audio quality, AFAIK |
| 23:01.48 | kn0x | kc8pxy: LSB=disstro? |
| 23:02.55 | kc8pxy | kn0x: lsb_release is a program that basically tells you what distro is running. |
| 23:03.13 | kc8pxy | kn0x: I'm no expert, but that's my first thought. |
| 23:03.17 | *** join/#asterisk telecos (n=sergio@42.166.219.87.dynamic.jazztel.es) |
| 23:04.00 | kami- | kc8pxy: yes, thank you. I just converted the wav to ulaw and have the same effect. But, as I'm not a developer of the system, Ihaven't heard the original file. I think the simplest thing would be to first listen to it. |
| 23:04.28 | kami- | it might be _really_ of bad quality |
| 23:07.21 | *** join/#asterisk kami- (n=user@unaffiliated/kami-) |
| 23:10.29 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:13.40 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:26.40 | KyleK | hrm toll free calls coming in from canada are expensive |
| 23:27.08 | drmessano | Well, yes |
| 23:28.30 | *** join/#asterisk unixdawg (n=unixdawg@c-76-125-161-44.hsd1.pa.comcast.net) |
| 23:28.44 | unixdawg | ok having a issue with a stand alone pbx |
| 23:29.10 | unixdawg | I registers 4 phones 1.6.0.5 and all of then are getting tos/cos errors |
| 23:29.15 | KyleK | drmessano: is this because most ITSP's are in the states? all I want is calls from canada :-/ |
| 23:29.17 | unixdawg | with what ever you dial |
| 23:29.33 | unixdawg | and then a fast busy |
| 23:30.20 | unixdawg | does not happen on 1.4.21 box but does on the 1.6.0.x |
| 23:31.12 | drmessano | KyleK: No, its because Canada is another country |
| 23:31.14 | drmessano | Period |
| 23:32.06 | unixdawg | les.net |
| 23:32.15 | unixdawg | look at them |
| 23:32.43 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 23:33.36 | KyleK | drmessano: well im in canada so its not another country from my perspective |
| 23:34.27 | drmessano | Who is the provider? |
| 23:35.54 | KyleK | well im looking for a provider right now |
| 23:36.43 | drmessano | If the providers are located in country X, any calls that are long distance to country X are subject to long distance charges |
| 23:36.46 | drmessano | Its that simple |
| 23:38.08 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:38.39 | KyleK | so _YES_ most ITSP's im finding on google are in the states, and thats why calls from canada are more |
| 23:39.03 | drmessano | Has nothing to do with where most of them are.. only the one YOU PICK |
| 23:39.12 | drmessano | So _NO_ |
| 23:40.16 | KyleK | you're either being very pedantic or not even reading what im saying :-( |
| 23:41.00 | drmessano | [19:29] <KyleK> drmessano: is this because most ITSP's are in the states? |
| 23:41.52 | KyleK | and with most ITSPs in the states, most ITSPs would charge for CAN -> USA call traffic |
| 23:41.59 | drmessano | No, has nothing to do with the location of the majority of the ITSPs.. The cost of a tollfree call from Canada is a function of the specific provider. Prevailing market statistics do not apply here. |
| 23:43.03 | KyleK | they do for figuring out that i need to search more specifically |
| 23:44.18 | drmessano | Not at all.. A provider in Saudi Arabia is going to charge an ass for a tollfree call from Canada.. but not because the majority of providers are in the USA. So your deductive reasoning is flawed here. |
| 23:45.25 | KyleK | my deduction was "i keep looking at USA providers and I really want a Canadian one specifically even though the numbers look the same" |
| 23:45.40 | drmessano | Thats not a deduction, that's a statement |
| 23:46.11 | KyleK | well its the result of my guess |
| 23:49.46 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 23:50.19 | KavanS | [May 3 16:48:44] NOTICE[3713]: chan_sip.c:13885 handle_request_invite: Call from '' to extension 'numberhere' because extension not found |
| 23:50.41 | KavanS | why is the first "from" entry blank? asterisk 1.4.18 |
| 23:50.54 | KavanS | trying to SIP trunk two asterisk boxes together, testing dial |
| 23:51.13 | KyleK | asterisk isn't seeing the incoming username as a user of the pbx |
| 23:53.02 | KavanS | hrm, why could that be? |
| 23:53.15 | KavanS | using a pap2 |
| 23:53.18 | *** join/#asterisk outtolunc (n=me@32.176.31.235) |
| 23:53.31 | KavanS | connected to asterisk, which is trunked to another asterisk box |
| 23:54.44 | KyleK | probably want to see what the incoming user name is based on the sip packet, might be like pap2@asteriskA |
| 23:56.48 | KavanS | KyleK, http://www.pastebin.ca/1411661 |
| 23:56.56 | KavanS | that's debug from the other server |