| 00:00.35 | coppice | its spelt 點心 |
| 00:01.40 | coppice | most people love dim sum places |
| 00:01.58 | coppice | the japanese go crazy for it |
| 00:02.20 | nestAr | DIM SUM! |
| 00:02.55 | Qwell | coppice: that was where I was told I needed to have chicken feet, blood pudding, and various other things that I declined |
| 00:02.55 | *** join/#asterisk |omni| (n=rob@c-67-185-193-76.hsd1.wa.comcast.net) |
| 00:04.30 | obnauticus | Hey is anyone ohere on Voipjet able to get their MD5 hash (PBX setup page)? |
| 00:04.41 | obnauticus | coppice, chiiinee? |
| 00:05.00 | BBHoss | obnauticus: works fine with me, but ive been a member for like 5 years |
| 00:05.03 | coppice | I remember going to a tea house in austin and having no idea what the things on the menu were. then we asked for a chinese version of the menu, and were OK. :-) |
| 00:05.15 | obnauticus | BBHoss that is interesting :| |
| 00:05.20 | obnauticus | Because I cannot get mine :\ |
| 00:05.25 | obnauticus | But I am a new member |
| 00:05.40 | BBHoss | how long ago did you sign up |
| 00:05.48 | obnauticus | I think my account is about |
| 00:05.54 | obnauticus | 4 months old :| |
| 00:06.02 | BBHoss | hmm |
| 00:06.04 | obnauticus | the number isn't that high though |
| 00:06.15 | obnauticus | I sent them an email.. |
| 00:06.34 | obnauticus | But fastresponse, isn't being so fast I sent it 2 days ago |
| 00:06.35 | obnauticus | i think |
| 00:06.42 | BBHoss | yeah im a 5XXX number |
| 00:06.57 | obnauticus | VoipJet account number (username/UserID): 169xx |
| 00:07.18 | obnauticus | I can use their nac server |
| 00:07.33 | obnauticus | BBHoss is it an image? |
| 00:07.44 | obnauticus | Maybe i can copy the URL structure of the image and get it for me |
| 00:07.45 | BBHoss | no, text |
| 00:07.48 | obnauticus | Oh |
| 00:07.50 | obnauticus | That's interesting |
| 00:07.55 | obnauticus | I tried generating it myself |
| 00:07.56 | BBHoss | no its just setup.php |
| 00:07.58 | obnauticus | and it didn't work out too well |
| 00:08.07 | obnauticus | Authorization code (password): (You should see an MD5 string, if it is blank logout and login again and/or use a recent version of Internet Explorer browser) |
| 00:08.55 | Qwell | coppice: the place I went, I couldn't tell what any of the stuff was.. so I had to rely on people to tell me. |
| 00:09.05 | Qwell | I wouldn't be able to go there by myself - it just wouldn't work |
| 00:09.15 | Qwell | there were no menus, just carts and food |
| 00:09.22 | BBHoss | what browser are you using |
| 00:09.23 | obnauticus | BBHoss I also tried using plaintext |
| 00:09.26 | obnauticus | and that didn't work |
| 00:09.35 | BBHoss | try using opera |
| 00:09.46 | BBHoss | it works fine in FF, but you never know |
| 00:09.47 | obnauticus | I'll try using a proxy from my webserver |
| 00:09.59 | coppice | Qwell: oh, nice. that's the real traditional style. they are disappearing fast. even if you don't like the food much, the atmosphere is fun |
| 00:10.08 | Qwell | the food was quite good |
| 00:10.25 | Qwell | I just have no idea what half of it was, heh |
| 00:11.36 | coppice | in most places you now tick items on a sheet, and they bring them to you. i long for the good old days |
| 00:12.02 | Qwell | coppice: she ticked it after she handed it to us |
| 00:12.25 | Qwell | the other difficult part there, was that the prices were quite unknown |
| 00:12.50 | Qwell | (it ended up being quite expensive, actually...) |
| 00:12.58 | coppice | ah, that's the billing side of the process. stamping the card with little chops is the real traditional way |
| 00:13.06 | Qwell | ahh, yeah, it was stamps |
| 00:13.23 | coppice | the pricing is usually quite clear |
| 00:13.33 | Qwell | with multiple lines for pricing, and just stamping the different lines |
| 00:13.50 | Qwell | coppice: well, I'm sure it was to most of the people there :) |
| 00:14.21 | Qwell | but that's okay - I wasn't paying anyways |
| 00:14.29 | coppice | the card usually has 3 or 4 areas, and the dishes are grouped into low medium and high price. they should stamp the appropriate section |
| 00:14.33 | Qwell | it ended up being like $20/each |
| 00:15.17 | coppice | you should leave the abelone and lobster alone :-) |
| 00:15.18 | kanatu | Can anyone help me with getting blind transfer to work on asterisk 1.2.21.1? |
| 00:15.26 | Qwell | coppice: we didn't get anything like that |
| 00:15.30 | Qwell | ...maybe :D |
| 00:15.36 | Qwell | there were just a lot of stamps |
| 00:16.01 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 00:17.31 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 00:26.50 | mamep | how can use opal with asterisk? |
| 00:29.28 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-d94fa9adf6084b2f) |
| 00:32.01 | jer | anyone have any documentation that i can read that will give me instructions to how i can enter a code to switch between the IVR and a ring-all scheme? i'd like to have, when i'm out of the house, the IVR, and when i'm back in the house, a ring-all situation |
| 00:37.04 | *** join/#asterisk SM0TVI (i=teodor@c80-216-203-230.bredband.comhem.se) |
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| 00:48.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:50.52 | codazoda | I have a machine that seems to lose the zaptel cards every time it reboots. Now, I'm trying to run "modprobe zaptel ; modprobe wctdm" to bring up the TDM404B card (I also have a TDM808B in the system, which I'll load second). When I run that, I get "line 0: unable to open master device '/dev/zap/ctl'". |
| 00:51.08 | codazoda | Is that a normal error since I haven't loaded the wctdm24xx module yet? |
| 00:53.57 | *** join/#asterisk shido6 (i=shido6@74-130-126-198.dhcp.insightbb.com) |
| 00:54.17 | codazoda | Maybe has something to do with udev. I do have udev on this system, but I build zaptel with 'make udev'. |
| 00:54.36 | codazoda | sorry, 'make install-udev'. |
| 00:55.45 | *** part/#asterisk codazoda (n=chatzill@c-67-186-201-157.hsd1.ut.comcast.net) |
| 00:55.51 | *** join/#asterisk codazoda (n=chatzill@c-67-186-201-157.hsd1.ut.comcast.net) |
| 00:57.19 | [hC] | Anyone aware of a parking bug in 1.2.18 where parking a second call makes the first call hang up, claiming a "No/unknown event '0' on timer' and 'I sh ould never be called! hanging up' ? |
| 01:02.24 | puzzled | [hC], not aware of one but if it's a bug I don't think it is going to be fixed since 1.2 is EOL |
| 01:02.54 | [hC] | puzzled: doesnt seem to happen in 1.2.19 so im gonna try upgrading these guys to that to see if it goes away first. |
| 01:05.19 | *** join/#asterisk crashsys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 01:06.32 | crashsys | Anyone ever tried to route a BRI across asterisk? Trying to take a BRI polycom video conf. phone and pass it out a PRI. |
| 01:06.47 | JT | asterisk doesn't support NA NRI |
| 01:09.29 | JackEStorm | JT: asterisk doesn't support alot of what it should. |
| 01:12.06 | [hC] | This is amusing. If my default route on my asterisk box is dead, (yet i have a static route to my one and only IAX peer for connectivity) i start to get a bunch of weird symptoms like phones ringing that you cant answer, or outgoing calls not dialing, and transfers failing. (all SIP phones, local LAN) |
| 01:12.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:13.06 | jer | cross connect to your provider? |
| 01:13.31 | JT | JackEStorm: heh |
| 01:13.35 | [hC] | what? |
| 01:14.02 | JackEStorm | JT: my biggest issue is the lack of sane clocking |
| 01:14.09 | JT | true |
| 01:14.15 | jer | [hC], nm |
| 01:14.45 | jer | anyone have any documentation that i can read that will give me instructions to how i can enter a code or dial an extension (or something) to switch between the IVR and a ring-all scheme? i'd like to have, when i'm out of the house, the IVR, and when i'm back in the house, a ring-all situation |
| 01:14.56 | [hC] | seems like asterisk's SIP code is wacky in some circumstances.. |
| 01:15.49 | JackEStorm | [hC]: not really, 1.2 used the routing table for src ip on systems with multi-interfaces and ips |
| 01:17.01 | JackEStorm | JT: it really makes you wonder what mark was thinking when he decided that frame clocking should be done this way. |
| 01:17.59 | [hC] | JackEStorm: so if all the sip clients are on, say, eth1.. and its available... if my eth0 link vanishes (which was also what my default route used), yet my iax peer has a static route over eth2, and its up, why should losing my default route start jacking up SIP devices? |
| 01:18.03 | [hC] | Ive seen this more than once |
| 01:18.24 | JT | JackEStorm: thinking? |
| 01:18.45 | [hC] | even.. when default route vanishes, and calls start coming in to Zap, sip devices behave erratically |
| 01:19.06 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 01:20.56 | JackEStorm | JT: asterisk has no internal clocking support, or support for clock pools, right now if my PRI goes down again, noone can check voicemail or dial out over failover, because of the lack of a clocking source. |
| 01:21.47 | JackEStorm | [hC]: yup, btw, are you proxy_arg on eth2? (in linux) |
| 01:21.56 | JackEStorm | err prox_arp |
| 01:21.58 | JackEStorm | shit |
| 01:22.19 | JT | JackEStorm: yes it's sily |
| 01:22.21 | JT | silly |
| 01:23.26 | [hC] | JackEStorm: I am not sure, id have to look, but i dont see how that has anything to do with how asterisk starts messing with local SIP devices because another interfaces route stopped passing traffic |
| 01:23.37 | codazoda | Crap. I'm working with a system that has been working fine for months. It was rebooted today and Zaptel no longer works. When I run ztcfg I get "line 0: Unable to open master device '/dev/zap/ctl'". Any idea what this means? |
| 01:24.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:25.17 | JackEStorm | [hC]: in linux, eth0 is the arp master, and in * 1.2, it uses the routeing table to build the outgoing packet headers. |
| 01:27.26 | JackEStorm | [hC]: I had a problem like yours, and after I enabled proxy arp on all interfaces, and switched to 1.4 it went away. |
| 01:27.59 | [hC] | JackEStorm: okay.. so if i start asterisk with the iax2 trunk being on eth0, then i quit asterisk, static routed the iax2 peer out over eth2, reconnected to asterisk and did an iax2 reload. it registered over eth2. THEN, eth0 goes away and everything starts flipping out |
| 01:28.25 | [hC] | JackEStorm: unless IAX2's packets are being constructed improperly and thus jacking up SIP... cause the SIP packets always go to the phones on the LAN, and that interface is always fine and present. |
| 01:31.41 | JackEStorm | [hC]: what ip blocks are on 0,1,2? |
| 01:32.39 | JackEStorm | (in CIDR) |
| 01:34.04 | [hC] | JackEStorm: I dont see how that matters. but lets say ADSL (eth0) is 200.1.0/24 and LAN )eth1) is 10.2.0.0/24 and WAN backup (eth2) is 77.50.1.0/24 |
| 01:34.51 | codazoda | okay, weird. If I load the wctdm24xxp BEFORE the wctdm, it works fine. Is there a way I can control what order these load in on bootup? |
| 01:37.18 | JackEStorm | [hC]: and in sip.conf you have localnet=10.2.0.0/24 and externip=?? |
| 01:38.01 | [hC] | JackEStorm: no, they are not set. They dont need to be set. |
| 01:38.13 | [hC] | I never do SIP on the WAN, on this box |
| 01:40.25 | JackEStorm | ok then, a echo 1 >/proc/sys/net/ipv4/conf/eth2/proxy_arp should solve it. |
| 01:40.37 | JackEStorm | (at boot, and now) |
| 01:41.30 | JackEStorm | (and I proxy_arp on all interfaces) |
| 01:41.47 | JackEStorm | eth0 is the only interface that proxies arp by default |
| 01:42.14 | [hC] | JackEStorm: Proxy arp doesnt seem to have much relevance to this... unless.. asterisk is caching something to do with interfaces/routing table at startup, most specifically in my case for IAX2... then if the routing table changes/paths become unavailable, IAX2 starts messing up, which in turn messes up SIP, which rears its head in the form of erratic behavior on desk phones |
| 01:42.28 | [hC] | JackEStorm: and the proxy arp would simply allow asterisk to arp out the now screwed up interface, as though it was still okay? |
| 01:43.43 | JackEStorm | [hC]: 1.2 used the routing table to construct out going packets., you are still using 1.2 |
| 01:43.54 | [hC] | JackEStorm: this is true. |
| 01:44.12 | [hC] | JackEStorm: and presumably cached this information on startup, instead of per-call.. |
| 01:44.46 | JackEStorm | the kernel has it cached |
| 01:44.55 | JackEStorm | and it pulls from the kernel |
| 01:45.55 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 01:45.59 | JackEStorm | and a echo 1 >/proc/sys/net/ipv4/conf/eth1/proxy_arp will help you out as well. |
| 01:46.17 | JackEStorm | (is is all assuming you are ip4) |
| 01:46.48 | [hC] | JackEStorm: OK, but clearly it was still doing something with regards to the default route, either in SIP, or in IAX, because SIP only communicates with LAN devices (nothing to do with the WAN, unless its doing it without my knowledge) and IAX is static routed over a different interface. From my perspective, i see no reason in this configuration why the default route needs to be addressed. |
| 01:47.13 | [hC] | JackEStorm: It must be something that SIP is doing (maybe in respect to not setting externip specifically) and trying to set it automatically? |
| 01:48.09 | Olobola | I keep forgetting: why isn't mp3 an option for voicemail? |
| 01:48.42 | jer | perhaps because it's encumbered? |
| 01:48.50 | JackEStorm | [hC]: back to that 'linux default interface is eth0' |
| 01:50.05 | JackEStorm | [hC]: with eth0 down/down the kernel lost it's arp cache. |
| 01:50.28 | Olobola | Too much CPU time to convert to mp3? |
| 01:50.30 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:51.05 | JackEStorm | Olobola: compression has patents, decompression doesn't |
| 01:51.45 | Olobola | oh, I see. |
| 01:51.52 | Olobola | thanks |
| 01:54.12 | JackEStorm | [hC]: you said that localnet is also unset? for sip? |
| 01:54.45 | JackEStorm | what is the bindaddr? |
| 01:54.57 | [hC] | JackEStorm: Im not sure i agree that linux does anything with regards to a "default interface" being set to eth0. That would ruin a lot of peoples configurations. |
| 01:55.27 | [hC] | JackEStorm: I do not set localnet, i do not send externip, and bindaddr=0.0.0.0 which clearly needs to be changed to my static SIP interface |
| 01:55.35 | [hC] | its set as such INCASE i need to connect a SIP device from the WAN |
| 01:55.50 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 01:55.55 | [hC] | but currently, for this client, they do absolutely ZERO sip on the WAN, and it all stays on the LAN. On the WAN they use only IAX. |
| 01:56.27 | [hC] | JackEStorm: it would seem as though since i set bindaddr=0.0.0.0, it is trying to guess what to set externip to, and picking an interface different from the one id like it to (LAN) |
| 01:56.41 | [hC] | when WAN disappears, it starts constructing packets improperly |
| 01:57.18 | JackEStorm | [hC]: with arp and eth0, it's always been that way, about 3months after SLS released the module patch for 0.85ish (can't remember) proxy arp was added. |
| 01:58.06 | [hC] | JackEStorm: it just makes zero sense, why anything would tie an arp cache to a 'default' interface.. it just invalidates the purpose of arp altogether. And i have plenty of boxes that have no eth0, that work just fine. |
| 01:59.30 | JackEStorm | [hC]: no eth0? and yet eth1 and eth2 with out alias support? |
| 02:00.15 | [hC] | JackEStorm: ehhhh.. interface aliases are ethX:Y. by no eth0 i mean linux sees there is an eth0, but it has no IP, or config on it at all. I simply used eth1. |
| 02:00.56 | JackEStorm | [hC]: btw, you never answered when eth0 is "lost" is it down/down? or up/down? |
| 02:01.13 | [hC] | JackEStorm: it never loses link. it just has no internet connectivity anymore. |
| 02:01.55 | JackEStorm | [hC]: you can module alias eth0 to eth50000 if you wanted, but the first loaded interface gets arp |
| 02:02.53 | JackEStorm | ethX:Y are ip aliases, not interface aliases |
| 02:05.23 | [hC] | JackEStorm: ok so let me see if i understand you correctly. You mean to tell me that if i have no ethernet cable plugged into eth0, and no IP set, and am solely using eth1 on a machine... all the arp traffic would be visible if i did a tcpdump on eth0? |
| 02:05.32 | [hC] | JackEStorm: and in the kernel arp cache, it would be stored as though it was received on eth0? |
| 02:05.42 | [hC] | JackEStorm: thus completely invalidating the purpose of an arp cache? |
| 02:07.00 | JackEStorm | [hC]: no, but the kernel will not send arp data to eth2 from eth1, only to eth0 but because it's down the link ignores it. |
| 02:07.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:07.55 | [hC] | JackEStorm: ok.. so then if i am trying to communicate between eth2>eth1, it will route, but if i try to communicate between eth2>eth0 it will not need to route? |
| 02:10.04 | JackEStorm | the src MAC and requested IP dest, are sent to the proxy_arp interfaces (and eth0 is one by default) |
| 02:12.12 | *** join/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 02:12.41 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 02:12.45 | [hC] | even though proxy_arp in /proc for eth0 is set to 0? All this means is that eth0 is defaulted to use proxy arp then, not that ALL arp is "forced" to eth0. all proxy arp allows is for another interface to answer arp on behalf of another interface |
| 02:13.23 | [hC] | the inherent problem here seems to be that either IAX or SIP packets are being generated with bad data, and locking/timing out if that data is bad, and the ip stack of box causes it to have to timeout |
| 02:13.58 | JackEStorm | [hC]: back to the 1.2 to 1.4 packet header issue |
| 02:14.05 | [hC] | for example, the SIP packet from * to the phone. the phone is registered via an IP address local to eth1.. Therefore, a packet TO this phone should use the IP of the * box on that interface, to the destination of the phone |
| 02:14.09 | [hC] | No other address. |
| 02:14.25 | [hC] | Oh, so this particular problem was specifically addressed in 1.4? |
| 02:15.31 | JackEStorm | yes, well no, more issues, but this one too, you can work around it with proxy_arp, but 1.4 is the solution, remember 1.2 uses the incoming IP to build the outgoing header |
| 02:16.06 | JackEStorm | and it pulls that from the routing table in the kernel |
| 02:16.40 | BillBinko | Sorry to interrupt... i'm having trouble posting to the asterisk-users group (tried from two addresses). This is making it hard to get help with my issue. I am getting the messages, but can't post. Am I missing something? |
| 02:16.51 | JackEStorm | (this is really a nat issue problem but if you proxy_arp on 1 and 2, and/or upgrade to 1.4 it's fixed) |
| 02:16.54 | [hC] | okay.. so the core of the issue here is that 1.2 crafts packet headers with bad presumptions |
| 02:17.03 | [hC] | and 1.4 is more sane, and should be used. |
| 02:17.03 | [hC] | :) |
| 02:17.14 | JackEStorm | [hC]: correct, BUT |
| 02:17.23 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 02:17.35 | [hC] | see why did you have to toss that BUT in there.. :) |
| 02:17.38 | JackEStorm | your config is hosed, proxy_arp, look in to it |
| 02:17.49 | [hC] | my config is hosed? |
| 02:18.08 | JackEStorm | not with asterisk, but with linux |
| 02:18.15 | [hC] | proxy_arp makes sense on natted interfaces, yes... so that another interface can answer on behalf of a now dead interface. |
| 02:18.21 | [hC] | And i am not doing any NAT. |
| 02:18.32 | [hC] | Nor should any interface have to respond for arp of a now dead interface. |
| 02:18.55 | *** join/#asterisk grimsy (n=chatzill@60-241-200-26.static.tpgi.com.au) |
| 02:20.18 | JackEStorm | [hC]: correct but eth1 still wants to send it to eth0 |
| 02:20.40 | *** part/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 02:20.41 | JackEStorm | [hC]: ok example, nothing todo with asterisk |
| 02:20.55 | [hC] | JackEStorm: I guess I just dont see why that would ever have to happen... I am not doing NAT here.. |
| 02:20.58 | *** join/#asterisk PC_Clone (n=pc_clone@c-76-16-79-154.hsd1.il.comcast.net) |
| 02:22.28 | JackEStorm | [hC]: eth0 goes to a router, eth1 goes to a router eth2 is the local lan, eth1 is a failover 100TX link of eth0 (not spaning tree, but T1 to ADSL failover) same IP's on both links |
| 02:23.32 | JackEStorm | when eth0 goes down, linux still thinks that eth0 is the arp master, for MAC's that don't respond. |
| 02:23.49 | JackEStorm | unless eth1 is set to proxy_arp |
| 02:26.00 | JackEStorm | [hC]: and eth1 is a router, and eth2 is a ADSL backup, same ip's, but eth0 goes to lan, it works. |
| 02:27.05 | JackEStorm | but you still need to proxy_arp in inbound traffic |
| 02:27.11 | *** join/#asterisk mtaht4 (n=m@161-106-62-200.enitel.net.ni) |
| 02:28.07 | JackEStorm | [hC]: this goes back to linux expecting only one device type ie: eth/wlan/ath....etc |
| 02:28.44 | *** part/#asterisk mtaht4 (n=m@161-106-62-200.enitel.net.ni) |
| 02:28.47 | [hC] | JackEStorm: okay your example makes sense because eth0 and eth1 are sharing the same IP space in the sense of a failover. |
| 02:29.09 | JackEStorm | look at /dev/hd? and /sec/sd? with sata now you will will start to understand. |
| 02:29.43 | [hC] | JackEStorm: ie both WAN links are addressed with the same addresses. in my case they are not. |
| 02:29.47 | JackEStorm | s%/sec/%/dev/% |
| 02:30.00 | PC_Clone | hello everyone, does anyone know if you can allow a user to forward a voicemail message to a user outside their context? |
| 02:30.59 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e9a2a50a99c4d0c7) |
| 02:35.18 | JackEStorm | [hC]: ok now, lets say that eth0 is 192.168.0.0/24, and eth1 is 192.168.1.0/24 and, eth2 is 192.168.2.0/24, and you have made sure that your distro is setup for classless |
| 02:36.15 | [hC] | JackEStorm: what do you mean by classless? |
| 02:38.06 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 02:39.32 | BillBinko | So, I just switched to a new server with 2 dual-core opterons. The quality on SiP calls is quite a bit lower than my ancient P3-400 I was using for testing |
| 02:39.47 | JackEStorm | [hC]: pre CIDR |
| 02:40.04 | BillBinko | I have no zaptel devices, so I'm thinking this is a clock issue? |
| 02:40.23 | [hC] | JackEStorm: I know what class based networks are, but i again I dont know what you mean by 'your distro is set up for classless' |
| 02:41.10 | JackEStorm | [hC]: no unintended network route? |
| 02:42.12 | JackEStorm | so, for lets say, 10.10.10.10 no netmask of 255.0.0.0? |
| 02:42.57 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.91) |
| 02:43.46 | JackEStorm | ie: using bitmasks and not byte masks? |
| 02:45.18 | [hC] | JackEStorm: all interfaces are configured using 255.255.255.0 (or according subnet) using ifconfig, not iproute2 |
| 02:45.54 | JackEStorm | and route -n? |
| 02:46.05 | [hC] | JackEStorm: but we're talking about completely different networks being used for routing here, not using one network, then failing off of an interface onto another, and then reusing the same addressing on the new interface. I am not doing that. |
| 02:47.34 | obnauticus | I wish i had a 3rd WAN so i could have dual wan + failover. |
| 02:47.37 | obnauticus | that would be pretty cool. |
| 02:55.40 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 02:57.16 | *** join/#asterisk _theHub (n=_theHub@ool-43577a99.dyn.optonline.net) |
| 02:59.40 | ViKing78 | How does the Aastra 57i compare to the Polycom 550? I like the programmable buttons on the 57i but does the audio compare to the Polycom? |
| 03:11.49 | *** join/#asterisk BeeBuu (n=root@218.13.87.243) |
| 03:12.01 | BeeBuu | hi,all |
| 03:12.49 | BeeBuu | why my read application doesn't work? |
| 03:13.01 | BeeBuu | in * 1.4.14 |
| 03:13.24 | *** join/#asterisk Nukemizer (n=Nukemize@15.249.sfcn.org) |
| 03:14.00 | BeeBuu | why my read application doesn't work in * 1.4.14? |
| 03:15.34 | BeeBuu | read(userinput|welcome|0|noanswer||30) |
| 03:16.01 | BeeBuu | the system just skip it,anyone help,please? |
| 03:19.20 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 03:19.40 | BeeBuu | hi,macros73 |
| 03:22.02 | JunK-Y | [hC]: BUMP! |
| 03:22.08 | [hC] | heyyyy junky! |
| 03:22.18 | JunK-Y | whats up? |
| 03:23.00 | BeeBuu | why my read application doesn't work in * 1.4.14? |
| 03:24.02 | [hC] | ahhh just trying to understand weird shit in asterisk, as usual :) |
| 03:24.53 | BeeBuu | :-( |
| 03:25.49 | [hC] | how are you doing? |
| 03:26.39 | JunK-Y | im great, lot of homeworks, but great, like usual :) |
| 03:26.40 | JunK-Y | you? |
| 03:27.50 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 03:28.02 | BeeBuu | hi,techle |
| 03:28.11 | [hC] | great :) busy as hell... cant wait for the next astricon.. that is the one time per year i get to associate asterisk and phones with nothing but fun :) |
| 03:32.21 | JunK-Y | damn ebay! |
| 03:36.01 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:39.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:48.56 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 03:56.32 | *** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net) |
| 03:58.26 | *** part/#asterisk BeeBuu (n=root@218.13.87.243) |
| 04:03.34 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 04:17.17 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:18.54 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:21.48 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 04:22.23 | *** join/#asterisk cmantito (n=gphreak@redbaron.kevinsnet.com) |
| 04:22.30 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:23.52 | cmantito | just a quick question, is it possible to authenticate a SIP phone/client connecting to a server based solely on it's IP address? |
| 04:24.35 | r0d3nt | to allow any sip client from a specific ip ? |
| 04:24.54 | obnauticus | Is there RADIUS authentication support with * ? |
| 04:24.58 | obnauticus | ../LDAP |
| 04:25.00 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 04:25.07 | cmantito | r0d3nt: yeah, I suppose |
| 04:25.35 | cmantito | more, I have an ATA that doesn't seem to want to send a registration correctly, but it uses a static IP so I wanted to see if I could kinda fudge it. |
| 04:27.19 | [TK]D-Fender | cmantito, I believe you can get away without having to auth the call for a pass, but I think it still has to match a name. |
| 04:27.58 | cmantito | bugger. well, this is what it seems to be doing: Registration from '<sip:*@10.13.36.1;transport=UDP>' failed for '10.13.36.150' - Username/auth name mismatch |
| 04:28.54 | cmantito | and it's entirely plausible that I may have configured something wrong, but all the softclients I've been using so far haven't had any issues yet. |
| 04:29.43 | [TK]D-Fender | cmantito, Whats the device? |
| 04:29.46 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 04:30.31 | cmantito | Linksys PAP2 v2 with (*gag*) Vonage's firmware on it, but it's been reprovisioned. I've got it to authenticate to other SIP services (Stanaphone, Demon VoIP), but so far not to any of my asterisks. |
| 04:31.04 | tripps | what is the proper way to debug dial-by-name directory? it doesn't seem to work (never finds an entry) though voicemail.conf has names in it. At CLI with verbose to 10 but don't see what keys are being pressed or why it's failing |
| 04:32.07 | [TK]D-Fender | cmantito, Should be pretty straightforward. fill in the password, and set "use authid=no. make sue in sip.conf you have [usenameenteredinPAP] and "secret=whatIenteredinto thePAP" |
| 04:32.22 | [TK]D-Fender | tripps, pastebin your dialplan and voicemail.conf |
| 04:32.33 | [TK]D-Fender | tripps, All applicable contexts. |
| 04:32.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:32.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:32.37 | [TK]D-Fender | ~pb |
| 04:32.37 | jbot | A Pastebin is a place to paste your stuff 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 |
| 04:32.42 | [TK]D-Fender | ^^^^^^^^^ |
| 04:32.54 | cmantito | [TK]D-Fender: does the authid bit get set in sip.conf? |
| 04:33.01 | cmantito | or is that something I'm supposed to set on the PAP2? |
| 04:33.18 | [TK]D-Fender | cmantito, You should set that to no in the PAP if that is the checkbox that it is in the higher models. |
| 04:33.49 | cmantito | I haven't found that on mine so far, I've been looking cause I read that somewhere else. You don't know anything else it might be referred to as? I don't see anything resembling it... :S |
| 04:33.50 | [TK]D-Fender | cmantito, thats a PAP2 setting I'm referring to. This IS a bit of an assumption that it carries that field from its big brothers |
| 04:34.06 | cmantito | that's what I was afraid of. |
| 04:34.19 | [TK]D-Fender | cmantito, If you can host screen shots or soemthing it might help, and pastebin your sip.conf entry for it. |
| 04:34.29 | cmantito | certainly. |
| 04:36.55 | cmantito | sip.conf: http://pastebin.com/d4773db60 |
| 04:37.28 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 04:37.57 | [TK]D-Fender | cmantito, Where is * relative to [kevin]? |
| 04:38.32 | cmantito | that's just it -- '*' seems to be what the PAP2 is trying to register as, despite the fact I've asked it to register as 'kevin' in the config. |
| 04:39.16 | cmantito | !mkdir webdocs/vonage |
| 04:39.27 | cmantito | ...wrong syntax, sorry |
| 04:39.28 | [TK]D-Fender | cmantito, Sorry, where is ASTERISk relative to [kevin]? |
| 04:40.17 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 04:40.38 | cmantito | oh, duh. my bad :P kevin is an ATA on the same subnet as the asterisk box, theoretically speaking, kevin makes calls out through sip-outgoing and receives them from the registered sip provider, as well as being able to use extnesions in extensions.conf |
| 04:41.02 | ViKing78 | How does the Aastra 57i compare to the Polycom 550? I like the programmable buttons on the 57i but does the audio compare to the Polycom? |
| 04:41.15 | [TK]D-Fender | cmantito, Um.. if you're regin [kevin] to *, then ALL calls should be going through * |
| 04:42.04 | cmantito | sip-outgoing is defined in sip.conf on the asteirsk box, so yes, it would be going through asterisk. |
| 04:42.15 | [TK]D-Fender | ViKing78, IP 550 = over priced. Polycom has superior call handling, audio quality, and physical feel. Aastra has superior soft-keys, and cheap backlight |
| 04:42.35 | cmantito | http://redbaron.kevinsnet.com/~kevin/vonage/Picture%203.png http://redbaron.kevinsnet.com/~kevin/vonage/Picture%202.png and http://redbaron.kevinsnet.com/~kevin/vonage/Picture%201.png are the screenshots. |
| 04:42.38 | obnauticus | Huh |
| 04:42.43 | obnauticus | how do I get LDAP :| |
| 04:42.51 | obnauticus | For * |
| 04:42.56 | obnauticus | I don't have ldap.conf :\ |
| 04:43.38 | [TK]D-Fender | cmantito, http://pastebin.com/m705dbbea |
| 04:44.20 | ViKing78 | [TK]D-Fender: Thanks, The IP550 isn't overpriced when compared to most PBX keysets. |
| 04:44.26 | cmantito | oh, it's context should simply be 'default', do I need to explicitly set that? |
| 04:44.49 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-45-103.bchsia.telus.net) |
| 04:45.06 | [TK]D-Fender | ViKing78, No, I mean its overpriced against polycoms OTHER models. |
| 04:45.12 | obnauticus | http://www.funnyordie.com/videos/fa1420df1f |
| 04:45.14 | obnauticus | ROFL! |
| 04:45.21 | [TK]D-Fender | ViKing78, It has veirtually no place IMO. |
| 04:46.17 | ViKing78 | [TK]D-Fender: IP Keysets for my incumbent PBX are >$450. Digital keysets are about $180 |
| 04:46.27 | nestAr | heh |
| 04:46.30 | nestAr | i like the 550 |
| 04:46.43 | Qwell | get the 650 |
| 04:46.50 | Qwell | 550 is pretty pointless |
| 04:46.52 | [TK]D-Fender | ViKing78, IP550 offers very little over all lower models at twice the cost. |
| 04:47.16 | [TK]D-Fender | ViKing78, Thing is that the 550 is a dead end. You're better off spending a few more bucks on the 650 if you're considering it. |
| 04:47.33 | [TK]D-Fender | cmantito, And really double check your password. |
| 04:47.40 | nestAr | i've never actually used the 650, but i can't imagine that i'd actually need anything more. |
| 04:48.37 | [TK]D-Fender | I've used jsut about every model, and frankly ANY will do for 99% of uses. |
| 04:48.41 | cmantito | [TK]D-Fender: I'm positive the password is correct; I'm using it on a softphone. |
| 04:48.51 | cmantito | and I've added the context to the SIP config. |
| 04:48.55 | [TK]D-Fender | cmantito, Set it again on your ATA to be sure. |
| 04:49.12 | cmantito | okeydokey. |
| 04:49.17 | [TK]D-Fender | cmantito, Then apply your * changes and reboot the ATA |
| 04:49.43 | [TK]D-Fender | ViKing78, Lets try this way : What kind of call volume and handling needs is this user going to have? |
| 04:50.06 | cmantito | and it's still failing to reg. |
| 04:50.44 | [TK]D-Fender | cmantito, can you PB the last attempt |
| 04:51.02 | cmantito | which, the sip debug output? |
| 04:51.13 | cmantito | or just the reg error? |
| 04:51.25 | [TK]D-Fender | cmantito, heck both.... |
| 04:52.15 | ViKing78 | [TK]D-Fender: I'm just looking for a model/family to standardize on. I like the polycom's for everything but the customizable buttons. I like the 57i's for that one feature. |
| 04:52.43 | [TK]D-Fender | ViKing78, Yes, Aastra wins that hands down against every other option I've ever heard of. |
| 04:52.45 | ViKing78 | [TK]D-Fender: All soft keys with digital labels |
| 04:53.13 | [TK]D-Fender | ViKing78, There's an up & a down. What kind of use do you see them applying for and for what percentage of calls? |
| 04:54.12 | ViKing78 | [TK]D-Fender: Mostly, BLF lamps. Manager stations that want to keep an eye on 4-5 employees |
| 04:55.38 | cmantito | [TK]D-Fender: http://pastebin.com/m5ed8aa91 |
| 04:56.33 | ViKing78 | [TK]D-Fender: Our current system has 4 system keys and 8 programmable keys for the 12 button phone and we also have a 24 button phone that has 20 programmable keys. They can be used from BLF, speed dial, macros, etc. I need an IP solution that can replicate most of that. |
| 04:57.53 | [TK]D-Fender | ViKing78, How many managers watching? |
| 04:58.01 | [TK]D-Fender | ViKing78, vs ho many "users" |
| 04:58.07 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 04:58.54 | ViKing78 | [TK]D-Fender: It's usually about a 1/5 ratio |
| 04:59.15 | metfan2007 | Hi all!! how can I debug cdr connection to a mysql server, Asterisk only says that it cannot connect, but no why |
| 04:59.24 | [TK]D-Fender | ViKing78, tot counts? |
| 05:00.54 | ViKing78 | Ah, no. Let's say 10-15 Mgrs and 50-75 EEs |
| 05:01.24 | [TK]D-Fender | ViKing78, Yeah, thats a little tougher... |
| 05:02.03 | [TK]D-Fender | ViKing78, 4-5 would mean IP650's for them and more limitednumber of line keys for call handling. A little less natural, but each could watch 5, and use 1 line key for calls |
| 05:02.21 | [TK]D-Fender | ViKing78, You can support up to 8 per key on a polycom and leave 5/6 for BLF |
| 05:02.35 | ViKing78 | [TK]D-Fender: I could use op-panel but I'd like something hardware |
| 05:02.42 | BBHoss | you could get an aastra 480i or 57i, those have tons of keys (softkeys) you can use |
| 05:03.03 | [TK]D-Fender | ViKing78, Could be the Aastra is for you... |
| 05:03.07 | ViKing78 | [TK]D-Fender: Yeah, That's how I have my 330s setup now. I used the second line key for a VM key |
| 05:03.09 | tripps | [TK]D-Fender: using guis so you probably don't want to see that |
| 05:03.12 | [TK]D-Fender | ViKing78, how cheap have you found them? |
| 05:03.32 | [TK]D-Fender | (inexpensive) |
| 05:03.47 | BBHoss | also you can use the XML applications to change them on the fly, or customize for each user |
| 05:04.02 | ViKing78 | [TK]D-Fender: The Aastra 57i is $204 at telephonydepot |
| 05:04.14 | BBHoss | the 480i is cheap |
| 05:04.29 | BBHoss | (inexpensive) :) |
| 05:06.12 | [TK]D-Fender | ViKing78, They've got their minus, and I'll say from a handling standpoint they don't hold up to Polycom, but it would be substantially better for your visibilty. |
| 05:07.10 | [TK]D-Fender | ViKing78, And I could vouch for TD excellent price & service. They move fast. |
| 05:07.15 | *** join/#asterisk GuyOCanada (i=GuyOCana@75.155.220.205) |
| 05:07.25 | ViKing78 | [TK]D-Fender: Yeah, only polycom option is a 601/650 with an expansion. Big and expensive. Too bad, I _REALLY_ like the polycoms |
| 05:07.51 | PC_Clone | ViKing78: have you thought about using an operator panel on the PC's? |
| 05:08.17 | [TK]D-Fender | ViKing78, Well you could do without the expansion, but it'd be a little less natural in a way. Frankly though, the 57i is less expensive and better at handling your priorities |
| 05:08.21 | ViKing78 | Definitely, telephonydepot is the best. ifax, their sister company is also wonderful for Hylafax support. |
| 05:08.35 | [TK]D-Fender | ViKing78, I wouldn't feel too guilty about it. |
| 05:09.24 | BBHoss | you can write some VERY flexible apps for the 57i too |
| 05:09.39 | BBHoss | you can change any button on the phone with the XML apps |
| 05:09.54 | cmantito | aha...I got it to at least authentciate :D |
| 05:10.09 | GuyOCanada | Hello, |
| 05:10.16 | ViKing78 | PC_Clone: I could use op-panel but I'd like something hardware |
| 05:10.52 | ViKing78 | Thanks for the input guys. |
| 05:11.03 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 05:11.08 | [TK]D-Fender | cmantito, Great news |
| 05:11.23 | GuyOCanada | I was looking for some tollfree number providers but they all have really different prices, could anyone recommend me a good place |
| 05:11.32 | cmantito | turns out, one of the quirks of this pap2 is that it will only use a numeric username |
| 05:11.48 | [TK]D-Fender | cmantito, lol, who knew! |
| 05:11.55 | ViKing78 | BTW I finally got CNAM working on my PRI to my PBX. I'll pastebin the config if anyone is interested. |
| 05:11.56 | cmantito | the *@<ip addr> changed to <uid>@<ip addr> in the reg when I changed the uid to an integer |
| 05:12.00 | [TK]D-Fender | cmantito, craziness I tell you! |
| 05:12.38 | *** join/#asterisk BPrintz (n=x@CPE001195f54833-CM0012c90d08ee.cpe.net.cable.rogers.com) |
| 05:12.44 | ViKing78 | No I'll I need to do is hack libpri to work making a EECT/2BT on a QSIG switch |
| 05:12.49 | BPrintz | Hey, I just installed and ran asterisk and it doesn't seem to be accepting connections on port 5060, what would cause this? |
| 05:13.18 | [TK]D-Fender | ViKing78, I know you can do 2BCT on 5ESS so far, not sure about the rest. |
| 05:13.34 | GuyOCanada | BPrintz: basically your firewall rules: iptables |
| 05:13.55 | [TK]D-Fender | BPrintz, and enable SIP debug on * CLI and show us the failed attempt |
| 05:13.58 | BPrintz | I stopped IPTables ... |
| 05:14.07 | [TK]D-Fender | BPrintz, in a PASTEBIN |
| 05:14.08 | [TK]D-Fender | ~pb |
| 05:14.08 | jbot | A Pastebin is a place to paste your stuff 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 |
| 05:14.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 05:14.18 | BPrintz | The failed attempt doesn't even register, I tried a raw connect even from the local machine to it with no luck |
| 05:14.34 | ViKing78 | [TK]D-Fender: Give me some hints. I might be able to get it to work with the QSIG signaling. I know my PBX supports it |
| 05:14.37 | cmantito | BPrintz: anything in /etc/hosts.allow or /etc/hosts.deny? |
| 05:14.44 | russellb | BPrintz: how did you try to "raw connect" ? |
| 05:14.46 | ViKing78 | [TK]D-Fender: Thanks in advance |
| 05:14.51 | BPrintz | both are empty |
| 05:14.57 | [TK]D-Fender | BPrintz, if you have SIP debug enabled at CLI you should see the attempt. if you don't you have a completely seperate networking issue |
| 05:14.57 | russellb | BPrintz: have another SIP application running, like ekiga? |
| 05:15.16 | [TK]D-Fender | BPrintz, if you do thats a proble, |
| 05:15.39 | GuyOCanada | google ekiga |
| 05:15.47 | GuyOCanada | dont we have a google bot? |
| 05:16.04 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 05:16.07 | BPrintz | I run APF and iptables on the machine, but like i said I disabled both.... the strange thing is like i said I can't even do a raw connect on the port from the local machine |
| 05:16.08 | [TK]D-Fender | BPrintz, You'll need to set any other local SIP app to use a DIFFERENT port. If thats the case your app make have stolen 5060 before * could get its hands on it and chan_sip fails to load because of that |
| 05:16.24 | [TK]D-Fender | BPrintz, And what is a "raw connect"? |
| 05:16.33 | [TK]D-Fender | BPrintz, we need some details if were to help you |
| 05:16.57 | BPrintz | like just telnet / or use putty raw connect to the port on the machine |
| 05:17.00 | GuyOCanada | [TK]D-Fender: raw connect by saying that what he means is opening a telnet session |
| 05:17.01 | BBHoss | anybody else here see a similarity between the snom's web interface and the aastra's? |
| 05:17.13 | [TK]D-Fender | BPrintz, those are *TCP* apps! Asterisk SIP is **UDP* |
| 05:17.18 | BPrintz | ah |
| 05:17.30 | BPrintz | what other info do you need fender? |
| 05:17.55 | [TK]D-Fender | BPrintz, thats like trying to talk Cantonese to a Syrian! |
| 05:18.00 | BPrintz | yeah I know :P |
| 05:18.06 | BPrintz | didn't think that it was UDP, my bad |
| 05:18.19 | [TK]D-Fender | BPrintz, Well so far your tests are completely invalid... so show us something that SHOULD succeed that doesn't :) |
| 05:18.48 | BPrintz | well I try to connect my X-Lite to it, and I don't see any info come up on the console.... |
| 05:18.53 | BPrintz | running verbosity of 7 |
| 05:19.18 | [TK]D-Fender | BPrintz, so "sip debug" and see from there |
| 05:19.33 | BPrintz | then try and reconnect? |
| 05:19.44 | [TK]D-Fender | BPrintz, basic verbose won't tell you anything about auth failures, etc. |
| 05:19.59 | [TK]D-Fender | BPrintz, then try to reg again, then try to place a call,e tc. Pastebin the complete output. |
| 05:20.00 | [TK]D-Fender | ~pb |
| 05:20.01 | jbot | A Pastebin is a place to paste your stuff 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 |
| 05:20.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 05:20.13 | *** join/#asterisk lesterc (n=lesterc@221.199.209.30) |
| 05:21.17 | GuyOCanada | I Need a TollFree DID for canada |
| 05:21.51 | BPrintz | Fender, still dispalying nothing in my console, the X-Lite doesn't get past the attempt to register and making a call just times out |
| 05:21.51 | [TK]D-Fender | GuyOCanada, www.unlimitel.ca , www.les.net |
| 05:22.07 | GuyOCanada | i have a question about les.net |
| 05:22.17 | GuyOCanada | do you have an account with them? |
| 05:22.18 | [TK]D-Fender | BPrintz, try "module reload chan_sip.so" |
| 05:22.32 | [TK]D-Fender | BPrintz, and where is your X-lite relative to your * box? |
| 05:22.57 | [TK]D-Fender | GuyOCanada, I have experience with Unlimitel, but not Les. Others here have used them and say they're "OK" |
| 05:23.53 | BPrintz | Fender, reloaded it still same.... my X-Lite is on my machine, my Asterisk is running on a dedicated elsewhere ... |
| 05:24.03 | GuyOCanada | [TK]D-Fender: what is the monthly and setup prices of unlimitel |
| 05:24.35 | [TK]D-Fender | GuyOCanada, Go check out their site. |
| 05:24.59 | [TK]D-Fender | BPrintz, some networking details please. the Details can KILL you with VoIP..... |
| 05:25.05 | GuyOCanada | well it only shows 4cent per minute on their site |
| 05:25.24 | GuyOCanada | but there must be a DID number fee they cant get it for free can they |
| 05:25.38 | BPrintz | Fender what details do you need exactly, i've disabled the firewalls on the box ... |
| 05:26.20 | [TK]D-Fender | BPrintz, Does your * server have a public ip? how about your softphone? What routers if any, where? |
| 05:26.47 | [TK]D-Fender | GuyOCanada, just call them and find out. |
| 05:26.57 | BPrintz | Server has multiple public IPs .... softphone is behind a router, but I can connect to another asterisk server fine with it and place calls |
| 05:28.10 | [TK]D-Fender | BPrintz, pastebin your sip.conf and tell me what happened when you tried tor eload that module I told you to. |
| 05:28.14 | BPrintz | GuyOCanada try link2voip.com ... |
| 05:28.19 | BPrintz | k sec |
| 05:32.55 | BPrintz | http://pastebin.ca/790232 |
| 05:39.22 | BPrintz | Any ideas fender? |
| 05:40.02 | Mw3_ | hi. is this digium voicebus technology some kind of marketing trick or is it really improves the quality? |
| 05:40.56 | [TK]D-Fender | BPrintz, please permanently remove every commented out line in your sip.conf and repastebin. |
| 05:41.30 | GuyOCanada | voicebus? |
| 05:42.14 | *** part/#asterisk variable_office (n=variable@208.73.60.2) |
| 05:42.16 | Corydon76-dig | Mw3_: It's not a marketing trick |
| 05:42.21 | russellb | it's the pci to tdm chipset that the new digium cards use |
| 05:42.26 | russellb | that was developed at digium |
| 05:42.57 | GuyOCanada | never heard of it |
| 05:43.19 | Mw3_ | can i find somewhere information about it? (regarding why is it better than the former technology) |
| 05:44.04 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 05:45.02 | BPrintz | fender, are there any sample configs i can just grab easily? |
| 05:45.44 | [TK]D-Fender | BPrintz, flush the filler and lets see whats left. |
| 05:48.01 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 05:48.08 | metfan2007 | hey, I'm trying to make a call via a SIP provider, but when I hit the dial Asterisk prompts an audio "An error has ocurred, please contact technical support with error number 205" what is this??? |
| 05:48.49 | [TK]D-Fender | metfan2007, * doesn't make that errror, only some GUI you're running on top of it. |
| 05:49.06 | [TK]D-Fender | metfan2007, Pastebin the full SLI output of your failed call with SIP debug enabled |
| 05:49.09 | [TK]D-Fender | ~pb |
| 05:49.09 | jbot | A Pastebin is a place to paste your stuff 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 |
| 05:49.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 05:50.00 | BPrintz | http://pastebin.ca/790241 |
| 05:52.20 | [TK]D-Fender | BPrintz, add "nat=yes", "canreinvite=no", "disallow=all", "allow=ulaw" in that order to [bprintz]. Apply your changes, then enable SIP debug and test. |
| 05:53.21 | lesterc | guys - is the source extension get set into any magic variable like ${EXTEN} for the dailed number? |
| 05:53.24 | BPrintz | k sec, thanks |
| 05:53.55 | Corydon76-dig | You mean ${CALLERID(num)} ? |
| 05:54.18 | lesterc | Corydon76-dig: yeah that's it! :) |
| 05:54.23 | lesterc | cheers. |
| 05:54.41 | BPrintz | still nothing coming up on the console fender... the server isn't behind a nat either |
| 05:55.02 | *** join/#asterisk variable_office (n=variable@208.73.60.2) |
| 05:55.12 | variable_office | when compiling zaptel should it be done before or after asterisk? |
| 05:55.24 | Corydon76-dig | before |
| 05:55.45 | Corydon76-dig | installing zaptel should be done before compiling Asterisk |
| 05:56.16 | [TK]D-Fender | BPrintz, if your server isn't seeing anything then you have a networking issue. chan_sip is loaded. Seeing nothing tells you to look elseewhere |
| 05:56.44 | [TK]D-Fender | variable_office, Before |
| 05:56.44 | BPrintz | Fender, do you have a softphone that you can try connecting? |
| 05:56.50 | variable_office | thanks, that what I thought, but wanted to make sure it has been a while since i used zaptel |
| 05:56.59 | [TK]D-Fender | BPrintz, PM me the ip/user/pass |
| 05:57.20 | metfan2007 | [TK]D-Fender: http://pastebin.ca/790247 |
| 05:57.45 | metfan2007 | [TK]D-Fender: I'm not using any app on top of Asterisk |
| 05:59.54 | metfan2007 | [TK]D-Fender: And is Alison's voice... :S |
| 06:02.16 | [TK]D-Fender | metfan2007, call looks pretty normal there.... wheres the failure? |
| 06:03.44 | metfan2007 | [TK]D-Fender: Instead of contact the phone number, the voice prompt saying that an error ocurred appears, maybe is on the provider side? |
| 06:04.21 | [TK]D-Fender | metfan2007, *I* don't see it in your PB, do you? :) |
| 06:05.01 | [TK]D-Fender | metfan2007, So its looking like YOUR call is fine and THEY have issues :) |
| 06:05.12 | metfan2007 | [TK]D-Fender: hehe, ok, thanks |
| 06:06.22 | [hC] | Have any of you figured out a way to instruct a polycom phone (with no physical interaction with the phone) to join a particular VLAN? |
| 06:08.43 | [hC] | Other than CDP i mean. |
| 06:09.34 | PC_Clone | h |
| 06:09.46 | PC_Clone | [hC]: edit sip.conf? |
| 06:10.26 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 06:10.35 | [hC] | where's my trout.. |
| 06:11.50 | PC_Clone | thats not phyisical notification |
| 06:11.56 | PC_Clone | err interaction |
| 06:15.46 | [hC] | Well i just found that trixbox has vendor moderated forums |
| 06:15.51 | [hC] | which is interesting in itself |
| 06:16.07 | [hC] | but second, polycom posted a way to do what im looking for which is undocumented anywhere else, or in their example sip.cfg |
| 06:16.09 | [hC] | which boggles me |
| 06:16.13 | [hC] | since its so freaking big. |
| 06:17.23 | PC_Clone | look at the admin pdf where you got your sip firmware |
| 06:17.29 | PC_Clone | it really helps |
| 06:18.12 | [hC] | yes thanks. its not in there either. |
| 06:18.12 | [hC] | :) |
| 06:18.50 | PC_Clone | so are you looking for the lines you need to add to the sip config or something else? |
| 06:20.59 | [TK]D-Fender | PC_Clone, Do you actually have a clue about what he's asking for? This is WAY before thinking about SIP.... |
| 06:21.15 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e62642e514e98da1) |
| 06:21.20 | [hC] | Actually in his defense, i first thought he was talking about sip.conf for asterisk |
| 06:21.32 | [hC] | not realizing he meant sip.cfg for polycom config, and the setting is infact in the new admin guide |
| 06:21.41 | PC_Clone | I was talking about the sip.cfg file that you put int he ftp for polycom |
| 06:21.45 | [hC] | I just didnt find it until i knew what keyword i should search for. |
| 06:21.51 | [hC] | But they definitely dont make it clear |
| 06:21.58 | [hC] | I didnt even know there was such a thing as "device.set" commands |
| 06:22.11 | [TK]D-Fender | ok, its late.. I'm off... later all |
| 06:22.19 | [hC] | So now I guess i will go for the same approach as I do for aastra |
| 06:22.20 | PC_Clone | ouch |
| 06:22.45 | [hC] | provision once off of a "clean" provisioning directory with no phone configs in it, that simply says "please go to vlan X and reboot yourself" |
| 06:22.55 | [hC] | upon entering the new vlan, hits a different config server and pulls the appropriate config |
| 06:24.11 | PC_Clone | well the prob there might be that the dhcp (if you use it) will still point at the old server |
| 06:24.38 | [hC] | the whole idea behind moving it to a new vlan is so that doesnt happen |
| 06:24.42 | [hC] | ill give it a try and see what it does. |
| 06:24.59 | PC_Clone | ah....ya...it would hit that vlan's dhcp |
| 06:25.00 | PC_Clone | woops |
| 06:25.38 | obnauticus | That's what she said. |
| 06:26.02 | PC_Clone | no...she said "asshole, I"m not on birthcontrol" but that's a different story |
| 06:26.10 | obnauticus | ... |
| 06:26.14 | obnauticus | And that's how you were born. |
| 06:26.22 | PC_Clone | yep |
| 06:26.48 | *** join/#asterisk Mmurdock (n=vnjyjta@17.sub-72-121-30.myvzw.com) |
| 06:30.15 | PC_Clone | ok, I'm going to sleep |
| 06:30.18 | PC_Clone | later all |
| 06:33.45 | *** join/#asterisk Mc1ovin (n=TylersMo@12.206.234.39) |
| 06:36.01 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 06:42.44 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 06:48.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:50.03 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.45) |
| 06:50.50 | zeeesh | getting problem with voicemail.conf will anybody guide ? |
| 06:52.17 | *** join/#asterisk MicW (n=michael@dslb-088-074-155-083.pools.arcor-ip.net) |
| 06:52.21 | MicW | hi |
| 06:52.40 | zeeesh | i can hear my voicemail msg by using sip client but still could not send wave file in email, will anybody guide? |
| 06:53.20 | MicW | i run asterisk on my local server which gets a new ip every night. the problem is that after i got the new ip all external sip connections remain in the state "Request Sent". |
| 06:54.22 | MicW | i'm trying to debug this issue |
| 06:54.39 | MicW | asterisk sends a lot of "register" messages to the server |
| 06:54.56 | MicW | the server always responds with "SIP/2.0 401 Unauthorized" |
| 06:56.02 | MicW | the problem seems to be the digest parameter "nonce" which is always the same in asterisk's request |
| 06:56.08 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:56.11 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:57.06 | MicW | after i run "reload", asterisk responds to the "unauthorized" responses with a new registration message with the same "nonce" value as it got from the server |
| 07:00.11 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 07:00.49 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:03.01 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:05.59 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:07.30 | endre | good morning all |
| 07:07.35 | endre | what a beautiful day it is |
| 07:08.09 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 07:08.55 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 07:11.32 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:15.29 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 07:15.49 | MicW | endre: -2°C and clouds... |
| 07:16.54 | harpal | I have just configured asterisk. now which softphone is good? |
| 07:17.13 | zeeesh | <endre>: where is this beauty ... i m unable to c this .. coz i struck with my voicemail? |
| 07:20.00 | endre | zeeesh: you must be a bot, right>? |
| 07:22.58 | zeeesh | <endre>; so will u help .... i am unable to send my voicemail to any email address ? |
| 07:23.17 | tzafrir | ~bot |
| 07:23.17 | jbot | I ain't no stinkin' bot. I am a finely tuned and hand crafted tool. Oh wait... I guess I am a bot (that you should not abuse). |
| 07:23.23 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 07:25.19 | tzafrir | harpal, on what platform? SIP or IAX? |
| 07:25.35 | zeeesh | i can hear all my voice msgs by using sip client by i m unable to send these wave file to any hotmail or yahoo or any email address will anybody guide ... ? |
| 07:26.44 | endre | MicW: % mysql temp -e 'select round(avg(degc),2) from measures where time > unix_timestamp()-300 and probeid=1;' |
| 07:26.46 | harpal | tzafrir, I have just configured SIP |
| 07:26.47 | endre | 4.13 |
| 07:27.01 | endre | temperature outside, last 5 min moving average |
| 07:27.19 | tzafrir | zeeesh, and can you send any other mail message? |
| 07:27.35 | MicW | endre: *g* |
| 07:27.48 | tzafrir | echo test | mail -s test123 someone@yahoo.com |
| 07:28.07 | endre | zeeesh: no, sorry sir, im offtopic here |
| 07:28.17 | harpal | tzafrir, I have asterisk installed on debian machine. and I want softphone for windows. for SIP |
| 07:28.18 | endre | i've seen asterisk like 4 months before last time |
| 07:28.41 | tzafrir | Sorry, don't know windows clients... |
| 07:28.47 | zeeesh | <tzafrir>: i can hear all voice msgs by using xlite but i still could not send it to any email address |
| 07:29.07 | tzafrir | zeeesh, and can you send any other mail message? |
| 07:29.48 | zeeesh | no i can't |
| 07:29.49 | harpal | ok any one know which softphone is best for windows? |
| 07:30.22 | zeeesh | <harpal>: i m using xlite .. and its ok ... |
| 07:31.01 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:31.25 | zeeesh | <harpal>: http://www.counterpath.com/ |
| 07:32.14 | harpal | zeeesh, thanks. |
| 07:32.37 | *** join/#asterisk emk (n=emk@212.49.87.126) |
| 07:32.53 | harpal | I have configured my asterisk like http://www.pastebin.ca/784167. and how to connect softphones? I am new to this |
| 07:36.19 | tzafrir | zeeesh, so you need to fix your MTA installation |
| 07:36.38 | tzafrir | zeeesh, what linux distro? what MTA? |
| 07:36.50 | tzafrir | better ask at #$DISTRO |
| 07:36.54 | zeeesh | <tzafrir> rhel4 |
| 07:37.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:37.23 | tzafrir | sendmail? postfix? |
| 07:37.38 | tzafrir | Have you looked at the relevant logs? |
| 07:37.54 | zeeesh | in voicemail.conf servermail=asterisk ... here will i mention smtp .. and if i don't hv any smtp then what shud i do ? |
| 07:38.12 | tzafrir | I figure #redhat might be a better place to ask about this. As you can reproduce a problem without Asterisk involved |
| 07:40.30 | zeeesh | <tzafrir> i just want to ask ... where shud i give my smtp settings with username and password .. bcoz i could not get the place where i will mention these settings? |
| 07:40.47 | tzafrir | zeeesh, not with Asterisk |
| 07:41.02 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 07:41.03 | tzafrir | asterisk sends the message through /usr/sbin/sendmail |
| 07:41.19 | tzafrir | and that is sendmail, postfix or whatever |
| 07:41.42 | tzafrir | (or even a non-queuing MTA such as null-mailer) |
| 07:42.34 | zeeesh | <tzafrir> do i need to mention there any specific smtp or ... in voicemail.conf "servermail=asterisk" will work ? |
| 07:43.02 | ManxPower | zeeesh: no, you do not need to mention specific SMTP. |
| 07:43.11 | zeeesh | ok thnx |
| 07:43.18 | ManxPower | Asterisk will run /usr/bin/sendmail to handle the e-mail delivery. |
| 07:43.25 | tzafrir | zeeesh, the defaults should work (without setting anything in voicemail.conf) |
| 07:44.54 | zeeesh | ok thnx |
| 07:48.11 | *** join/#asterisk ronchilla (n=ronke@213.185.113.93) |
| 07:49.25 | ronchilla | Hello! i'm a newbie and i have a problem can anybody help? |
| 07:54.16 | tzafrir | ~ask |
| 07:54.16 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:55.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:01.28 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-150-56.dslextreme.com) |
| 08:06.45 | BezNalogov | I have big problems with my grandstream 2000 and connecting it to an asterisk server that has a public ip#. My firewall/router has very limited possibilities to configure. I do see an option in the grandstream to set a proxy. Could this be helpful to solve my problem? If so what kind of proxy is this and how can I setup it? |
| 08:07.00 | *** join/#asterisk dworkin (n=asdf@host86-144-40-253.range86-144.btcentralplus.com) |
| 08:08.50 | BezNalogov | I now have it working through a vpn tunnel btw, but the problem is that it isn't possible to select any menu items, for example the voicemail won't work. I get his error: |
| 08:08.53 | BezNalogov | [Nov 20 08:53:30] WARNING[10195]: chan_sip.c:1921 retrans_pkt: Maximum retries exceeded on transmission b060eca87eea6952@192.168.1.65 for seqno 25529 (Critical Response) |
| 08:08.53 | BezNalogov | [Nov 20 08:53:30] WARNING[10195]: chan_sip.c:1945 retrans_pkt: Hanging up call b060eca87eea6952@192.168.1.65 - no reply to our critical packet. |
| 08:08.53 | BezNalogov | <PROTECTED> |
| 08:12.04 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:14.20 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 08:17.38 | ronchilla | i'm running asterisk 1.4 on ubuntu server 6.06 using wtc410p card and get the ff error when i try to use the 2nd span |
| 08:17.49 | ronchilla | PRI got event: HDLC Abort (6) on Primary D-channel of span 2 |
| 08:18.22 | ronchilla | i've googled the issue and i've tried all the suggestions but none of them seem to work for me |
| 08:18.40 | ronchilla | to make matters worse i'm a linux newb too |
| 08:19.33 | ronchilla | can someone tell me what i'm doing wrong? |
| 08:19.37 | ronchilla | or not doing? |
| 08:21.42 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:29.58 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:30.25 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-580557cfdcc99370) |
| 08:33.39 | obnauticus | IS there any way to listento like a phone without it being `answered' |
| 08:33.50 | obnauticus | like I want it to automatically answer so i can like hear what's going on in that room. |
| 08:33.56 | obnauticus | (my room) |
| 08:35.35 | tzafrir | obnauticus, analog phone? |
| 08:35.48 | obnauticus | Cisco 7960 |
| 08:36.19 | obnauticus | I think I found it |
| 08:36.20 | obnauticus | http://www.voip-info.org/wiki/view/Cisco+7940-7960+auto-answer+config |
| 08:36.25 | obnauticus | Is that...what im looking for? |
| 08:42.36 | creativx | auto answer would be the keyword |
| 08:43.10 | obnauticus | ya |
| 08:43.51 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:50.49 | zeeesh | and then just change ... in xlite ... sip.proxy where u mentioned 5060 replace with 8891 .. and then try again ... |
| 08:52.34 | *** join/#asterisk Hyber[at]Work (n=Blah@firewall.rantek.dk) |
| 08:53.29 | Hyber[at]Work | anyone know a good VoIP/SIP client for Gnome/KDE? perferrable with multiple lines and quick dials. |
| 08:56.44 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 08:57.05 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2a66efa9156c0ec1) |
| 08:57.34 | cjk | hello, i use Goto() brefore doing Hangup() and like this the dst in my cdrs is always "h". any ideas or hints? |
| 08:57.44 | cjk | if i remove the Goto and do the Hangup immediately its ok |
| 08:57.52 | *** part/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) |
| 08:57.53 | *** join/#asterisk macTijn (i=martijn@rachel.insecure.nl) |
| 09:07.28 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 09:07.30 | phix | hey |
| 09:11.01 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:13.12 | MicW | when i set call redirection on my snom phone, a get an error in asterisk: No channel type registered for 'Local' |
| 09:15.03 | MicW | my mistake. i forgott t laod the "chan_local" module" |
| 09:29.23 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:30.01 | MicW | is there a way to pass commands to the asterisk console using a script? |
| 09:30.18 | MicW | this is not working: echo "command" | asterisk -r |
| 09:31.18 | creativx | asterisk -rx |
| 09:31.22 | ronr | asterisk -r -x "command" |
| 09:31.31 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:31.37 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 09:32.18 | MicW | oh, ok :) |
| 09:32.32 | MicW | much easier than i tried |
| 09:35.05 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 09:39.21 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:39.24 | Uatec | Hi |
| 09:39.54 | Uatec | does anybody know of a kind of "null" sip device? |
| 09:40.04 | Uatec | i'm developing a new asterisk implementation |
| 09:40.43 | Uatec | and I want to test my dialplan, and it would be much easier if i had some kind of app which, although it might not function like a phone, would Register and "ring" and "answer" as if it was actually a sip phone... |
| 09:40.46 | Uatec | or am i dreaming? |
| 09:41.46 | harpal | I have setup asterisk and now two xlite client I have connected. now when I try to call than it shows Call Failed. and in asterisk -r it shows me |
| 09:41.47 | harpal | chan_sip.c:13669 handle_request_invite: Call from '100' to extension '200' rejected because extension not found. |
| 09:47.23 | Uatec | harpal, you need to tell asterisk what to do when an extensions is dialed |
| 09:47.23 | Uatec | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 09:48.42 | harpal | Uatec, thanks |
| 09:49.04 | *** join/#asterisk anujsingh (n=anuj@59.90.65.14) |
| 09:49.11 | agx | Uatec, sipsak ? |
| 09:51.01 | anujsingh | is it possible to define rule based dialing with vicidial? asterisk, such as between 9 to 6 pm call on office , between 9 pm to 11 pm on cell and so on? |
| 09:52.23 | *** join/#asterisk ronchilla (n=ronke@213.185.113.93) |
| 09:56.15 | *** join/#asterisk blq (i=me@1.1.1.vg) |
| 09:56.30 | *** join/#asterisk bl4q (n=Bl@dslb-088-066-251-220.pools.arcor-ip.net) |
| 10:01.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:02.41 | Uatec | that is fantastically complicated! |
| 10:05.04 | *** part/#asterisk emk (n=emk@212.49.87.126) |
| 10:10.37 | *** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 10:16.38 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 10:18.56 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 10:30.21 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 10:38.54 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2a66efa9156c0ec1) |
| 10:53.27 | *** join/#asterisk und3r (n=und3r@213-140-22-122.ip.fastwebnet.it) |
| 10:53.32 | und3r | hi to all |
| 10:54.11 | und3r | i've a problem with update firmware of my grandstream gxp 2000 to version 1.1.4.18 |
| 10:54.20 | und3r | is there anyone can help me? |
| 10:54.34 | *** join/#asterisk macTijn (i=martijn@rachel.insecure.nl) |
| 10:54.45 | ManxPower | Grandstream should be able to help you. |
| 10:55.09 | und3r | i'm trying to update it from http on a local apache, i see that the telephone get the new file on boot but the firmware don't change |
| 10:55.12 | und3r | any idea? |
| 10:56.24 | ManxPower | und3r: Grandstream should be able to help you. |
| 10:56.52 | ManxPower | very few people here use Grandstream products. |
| 10:57.28 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:57.34 | und3r | ah ok, thanks anyway :/ |
| 10:57.45 | *** join/#asterisk skirmisha (i=skirmish@90-154-148-242.btc-net.bg) |
| 10:58.00 | agx | und3r, good luck with 1.1.4.x; i don't trust anything except 1.1.2.latest |
| 10:58.04 | skirmisha | any ideas why database put ignore spaces between values |
| 10:58.13 | skirmisha | agx how are u doing |
| 10:58.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:59.00 | agx | skirmisha, busy as hell :-( |
| 10:59.05 | skirmisha | asterisk -rx " database put cidname 1234 damn test" |
| 10:59.14 | skirmisha | and this gives me error |
| 10:59.23 | *** join/#asterisk bantu (n=Miranda@rz-du-fgx-146-116.rz.uni-karlsruhe.de) |
| 10:59.24 | skirmisha | damn test is ignored as there is space |
| 10:59.30 | skirmisha | how should i ad it |
| 10:59.59 | skirmisha | any ideas |
| 11:00.03 | agx | skirmisha, try escaping " with \" around damn test ===> \"damn test\" " |
| 11:00.18 | skirmisha | let me test |
| 11:01.07 | skirmisha | works |
| 11:01.12 | skirmisha | thanks a lot |
| 11:01.32 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 11:02.04 | mvanbaak | you can also mix double and single quotes |
| 11:02.18 | mvanbaak | asterisk -rx 'database put cidname 1234 "damn test"' |
| 11:05.14 | skirmisha | thanks |
| 11:08.24 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 11:09.36 | *** join/#asterisk n0s (n=n0s@c-76-103-43-96.hsd1.ca.comcast.net) |
| 11:13.57 | *** join/#asterisk gardo (n=gardo@121.97.143.36) |
| 11:15.53 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 11:18.19 | *** join/#asterisk falco_toadfoot (i=papir@104.80-202-255.nextgentel.com) |
| 11:18.56 | falco_toadfoot | How often are the qualify-messages sent? |
| 11:19.08 | falco_toadfoot | as in qualify=yes in sip.conf |
| 11:20.03 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 11:21.01 | hi365 | at what point are you supposed to install lib pri? z=zaptel, w=(sangoma)wanpipe a=asterisk aa=asterisk addons l=libpri ?? |
| 11:21.02 | agx | falco_toadfoot, 3000 i suppose |
| 11:21.14 | hi365 | z,w,l,a,aa? |
| 11:21.38 | hi365 | l,z,w,a,aa? |
| 11:21.43 | ManxPower | hi365: install zaptel, libpri, asterisk in that order |
| 11:21.57 | ManxPower | for sangoma, do whatever sangoma tells you to do |
| 11:22.18 | hi365 | thanks. do i need to stop the current services (asterisk, zaptel) to re-install? |
| 11:22.47 | ManxPower | hi365: no. |
| 11:22.54 | hi365 | thanks ManxPower |
| 11:23.31 | falco_toadfoot | agx: that seems strange, since I set the qualify=500 (wait 500 ms for response) and reloaded. Then I unplugged the sip client and waited at least 10 seconds before I got a PeerStatus event |
| 11:24.05 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:25.00 | ManxPower | falco_toadfoot: you cannot control how often the qualify is sent. |
| 11:25.48 | falco_toadfoot | ManxPower: ok, but how often are they sent? |
| 11:25.53 | BiG^DoG | All my extensions start with 2xx.... how do I get * to recognize I'm dialing an extension and not an external phone number so it completes the call faster? |
| 11:26.15 | ManxPower | falco_toadfoot: I would have to look at the source code. I suspect every 60 seconds or so. |
| 11:26.24 | agx | falco_toadfoot, peer status? ehm not that's the rate at wich asterisk send out "OPTIONS"... check chan_sip.c |
| 11:26.34 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 11:26.35 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:26.37 | ManxPower | BiG^DoG: that is why people use "9" to get an outside line. |
| 11:26.53 | ManxPower | you could also dial all PSTN calls as 1 + area code + number |
| 11:26.54 | BiG^DoG | ok, let's assume I don't want to do that |
| 11:27.08 | phix | hey |
| 11:27.24 | ManxPower | BiG^DoG: Asterisk CANNOT know if you are dialing extension 201 or phone number 2015551212 |
| 11:27.28 | phix | I would like to use ilbc but it seems like it has been taken out of the debian package :( |
| 11:27.29 | falco_toadfoot | ManxPower, agx: I'll check |
| 11:27.44 | phix | any one know of an addon to put it back in or do I need to recompile asterisk and set some flag? |
| 11:27.48 | ManxPower | now you can either use 9 for outside calls or somehow make sure there is no overlap between your extensions and PSTN numbers |
| 11:28.16 | BiG^DoG | well, we use 7 digit dialing for local calls and all long distance is 1+areacode+number |
| 11:28.23 | BiG^DoG | and there are no local prefixes starting with 2 |
| 11:28.33 | BiG^DoG | so I thought I could set up a dialplan (if that's the right word) for 2xx |
| 11:28.41 | ManxPower | BiG^DoG: then remove 2 from your starting pattern match |
| 11:28.49 | BiG^DoG | still learning the lingo |
| 11:28.52 | Uatec | short of using sipsak, which is waaaay too complicated for my purposes, does anybody know of a SIP emulator that I can use to test the connection and dialplans etc of multiple sip devices in my new asterisk implementation? |
| 11:29.06 | Uatec | i only need it to register and "Dial" and "ring" and "answer" |
| 11:29.07 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 11:29.32 | ManxPower | you would have something like exten => _NXXXXXX,1,Whatever So you would use exten => _[3-9]XXXXXX,1,Whatever |
| 11:29.47 | ManxPower | until you try to call a local number beginning with 2, then you are screwed |
| 11:30.20 | BiG^DoG | we don't have any local #'s starting with 2 right now! :) |
| 11:30.20 | JT | phix: iLBC is encumbered |
| 11:30.29 | BiG^DoG | so my outbound route right now is NXXXXXX |
| 11:30.38 | BiG^DoG | just replace that with 1XXXXXX, 3XXXXXX, etc? |
| 11:30.40 | JT | phix: so of course it's not in the debian package |
| 11:30.46 | ManxPower | no! |
| 11:30.52 | Uatec | i match 3 digit numbers agains internal extensions |
| 11:30.57 | ManxPower | did you miss the [3-9] in what I told you |
| 11:30.58 | Uatec | and 5 digit or greater numbers as external |
| 11:31.00 | Uatec | simple as |
| 11:31.09 | BiG^DoG | uatec, how? |
| 11:31.24 | ManxPower | Uatec: bigdog does not want to WAIT |
| 11:31.33 | ManxPower | He wants the call to start as soon as he is finished dialing |
| 11:31.48 | ManxPower | ManxPower: you would have something like exten => _NXXXXXX,1,Whatever So you would use exten => _[3-9]XXXXXX,1,Whatever |
| 11:32.14 | BiG^DoG | ah |
| 11:32.26 | BiG^DoG | and that's in extensions.conf, right? |
| 11:32.31 | Uatec | BiG^DoG: exten => XXX,1,Dial(SIP/${EXTEN}) and exten => _XXXX.,1,Dial(misdn/r:isdn/${EXTEN}) |
| 11:32.48 | ManxPower | yes, wherever you put your existing exten => _NXXXXXX, line. |
| 11:32.54 | Uatec | BiG^DoG, what kind of devices are you dialing with? |
| 11:32.57 | BiG^DoG | cool...thanks. |
| 11:32.59 | ManxPower | Uatec: you are missing the leading _ |
| 11:33.09 | phix | JT: encumbered? |
| 11:33.17 | phix | JT: hmmm it isn't even in the source package? |
| 11:33.31 | Uatec | oops, you're right ManxPower, thanks |
| 11:33.38 | Uatec | BiG^DoG, what devices are you using internally? |
| 11:33.46 | JT | phix: correct, it's encumbered, google it |
| 11:33.52 | JT | it's not a free codec |
| 11:33.54 | ManxPower | Uatec: you could have made BiG^DoG waste hours and hours of his life because of that little mistake. |
| 11:34.03 | Uatec | :( sorry |
| 11:34.11 | BiG^DoG | Uatec: Cisco 7940 |
| 11:34.14 | ManxPower | Uatec: BiG^DoG would have to be using Zap channels. |
| 11:34.18 | BiG^DoG | ManxPower: sorry, I'm not quite THAT stupid |
| 11:34.33 | ManxPower | BiG^DoG: for SIP phones, you would have to set all that stuff in the SIP phone, in addition to the dialplan |
| 11:34.35 | JT | BiG^DoG: you need to set the dialplan in the cisco accordingly too |
| 11:34.50 | BiG^DoG | k |
| 11:35.10 | ManxPower | BiG^DoG: SIP phones collect ALL the digits, when you are done dialing, then it sends the number to Asterisk all at once. |
| 11:35.37 | ManxPower | Asterisk doesn't even get notified when the SIP phone goes off hook or when you start dialing. |
| 11:36.58 | d-k-t | as long as you set cisco dialplan.xml to say 2xx are extensions, then it'll start as soon as you enter the 3rd digit |
| 11:37.14 | falco_toadfoot | anyone know where the ast_sched_add function is? |
| 11:37.25 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:37.28 | ManxPower | d-k-t: he wants to start dialing the 2xx extensions as soon as he is finished dialing the 3-digit extention |
| 11:38.30 | d-k-t | ManxPower, yep, which is why I said changing the cisco dialplan.xml would make that work :) |
| 11:38.54 | ManxPower | d-k-t: I'm sure BiG^DoG would appreciate help with that. |
| 11:39.06 | d-k-t | as long as there is no overlap overlap ofcourse |
| 11:39.13 | ManxPower | Since I don't use Ciscos (the licensing fees are way too high for my customers). |
| 11:39.34 | d-k-t | alternatively, you can dial a # to terminate number entry |
| 11:40.29 | d-k-t | not so intuitive to wives tho |
| 11:40.34 | d-k-t | or visitors |
| 11:41.52 | d-k-t | <TEMPLATE MATCH="2.." Timeout="0" User="Phone"/> would be the line you'd want in the dialplan.xml |
| 11:44.41 | d-k-t | probably want this to match the long distance ones too <TEMPLATE MATCH="1.........." Timeout="0" User="Phone"/> |
| 11:45.10 | ManxPower | how about for 7-digit local calls? |
| 11:45.48 | ManxPower | as well as 8-digit LD calls in the same area code and toll calls outside the local area code. |
| 11:47.37 | d-k-t | more of the same, . = X, timeout is how long it should wait before proceeding if you have overlapping matches for example |
| 11:47.46 | *** join/#asterisk Op3r (n=edwin@203.177.177.107) |
| 11:48.28 | ManxPower | he does not want to have timeouts. That is the entire reason he asked the question. |
| 11:48.50 | JT | timeout dialling sucks balls |
| 11:49.22 | ManxPower | JT: *nod* A well designed dialplan eliminates that for most people in the USA/Canada |
| 11:49.44 | d-k-t | ManxPower, basically then he needs to match his dialplan.xml to match his asterisk dialplan |
| 11:49.59 | JT | there is no reason why a no timeout dialling dialplan can be made for any country |
| 11:50.07 | JT | s/can/can't/ |
| 11:50.26 | JT | only international dialling may need to resort to timeout dialling |
| 11:50.31 | *** join/#asterisk myiagy (n=myiagy@201-67-138-60.bnut3703.dsl.brasiltelecom.net.br) |
| 11:50.44 | ManxPower | JT: many countries have variable length national dialplans |
| 11:50.55 | JT | that's no excuse |
| 11:51.11 | JT | people just need to learn some pattern matching skill ;) |
| 11:51.23 | d-k-t | and might make it somewhat longer |
| 11:51.48 | ManxPower | My dialplans are pretty simple. 4-digit extensions, 9+7 digits all calls in the same area code, 9+1+AC+number for all calls outside the area code. |
| 11:52.46 | ManxPower | *sigh* Today is going to be hell. In the middle of a major e-mail server upgrade. Now I'm just waiting for 88GB of mail to be transfered from the old server to the new server. |
| 11:52.50 | falco_toadfoot | ManxPower: line 11853 in chan_sip.c (1.20 branch) says: "this is done with the interval in qualify= option in sip.conf" |
| 11:54.07 | falco_toadfoot | so obviously one of the descriptions are wrong |
| 11:54.22 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 11:54.38 | falco_toadfoot | I'm guessing it's not interval |
| 11:55.05 | JT | falco_toadfoot: it's really quite simple, qualify= is the value of time in milliseconds before a host is determined to be unreachable |
| 11:55.25 | ManxPower | JT: He wants to know how often qualifies are sent. |
| 11:55.32 | JT | i know |
| 11:55.41 | JT | he still doesn't believe us as to what qualify= |
| 11:55.41 | ManxPower | and qualify= is not it. |
| 11:56.11 | ManxPower | qualify= specifies what the max response time is. Asterisk could be sending that request once per hour for all we know. |
| 11:56.21 | falco_toadfoot | JT: voip-info says two different things: that qualify=2000 means qualifies are sent every 2 seconds, and that qualify=2000 means that a sip_poke will expire after 2 seconds |
| 11:56.35 | JT | falco_toadfoot: well it needs editing then |
| 11:56.41 | ManxPower | the wiki is wrong |
| 11:56.41 | falco_toadfoot | ManxPower: exactly |
| 11:56.56 | JT | falco_toadfoot: just watch sip debug to determine the interval. |
| 11:57.08 | falco_toadfoot | ManxPower: also the comment for sip_poke_peer says that qualify= sets the interval |
| 12:00.00 | falco_toadfoot | JT: it seems it could be random |
| 12:00.37 | JT | falco_toadfoot: under a consistant set of conditions? eg, either always reachable or always unreachable? |
| 12:01.09 | falco_toadfoot | no, I meant that the interval could be random |
| 12:02.15 | cpm | I'm sorry. We want a block of flats, not an abattoir. |
| 12:03.18 | Uatec | ?? |
| 12:03.23 | coppice | but its a very high tech abattoir. no sprays of blood incommoding the passers by |
| 12:04.05 | JT | falco_toadfoot: i doubt it's random under constant conditions |
| 12:04.07 | cpm | Yes, and well done, but we wanted an apartment block! |
| 12:04.37 | cpm | nice as the abattoir is |
| 12:04.49 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128684591.dsl.bell.ca) |
| 12:05.21 | coppice | well it does have murals depicting Mediterranean scenes |
| 12:05.30 | coppice | just before the rotating knives |
| 12:08.02 | cpm | a gentleman of high refinement and culture indeed |
| 12:08.28 | coppice | Could you could put in a word for me. I'd love to be a mason. |
| 12:09.38 | yang | Could someone tell me the correct configuration for op_button (op-panel) http://openpaste.org/en/3986/ |
| 12:12.20 | falco_toadfoot | I found it: DEFAULT_FREQ_OK and DEFAULT_FREQ_NOTOK in chan_sip.c |
| 12:12.46 | falco_toadfoot | which is 10*1000 and 60*1000 respectively |
| 12:14.07 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 12:14.48 | cpm | it opens doors, I'm telling you! |
| 12:15.02 | rob0 | cpm is a bot, spewing Monty Python lines. |
| 12:15.40 | falco_toadfoot | there should be a qualify_freq option in sip.conf... |
| 12:18.39 | coppice | Well, of course, this is just the sort of blinkered philistine pig-ignorance I've come to expect from you non-creative garbage. You sit there on your loathsome spotty behinds squeezing blackheads, not caring a tinker's cuss for the struggling artist. |
| 12:21.52 | cpm | har! |
| 12:21.59 | cpm | very nice! |
| 12:24.14 | tzafrir | yang, for starters, it should be SIP/400 rather than SIP/16 |
| 12:27.47 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 12:27.50 | *** join/#asterisk mtaht4 (n=m@161-106-62-200.enitel.net.ni) |
| 12:28.17 | ManxPower | falco_toadfoot: why do you care? |
| 12:28.36 | ManxPower | What problem is caused by not knowing the qualify frequency? |
| 12:28.50 | falco_toadfoot | ManxPower: because I need to know before 60s have passed what the status is |
| 12:28.57 | falco_toadfoot | I need a shorter frequency |
| 12:30.23 | ManxPower | I can't imagine why. |
| 12:30.53 | ManxPower | what problem is caused by knowing if the device is unreachable in 58 seconds .vs. 62 seconds? |
| 12:31.27 | falco_toadfoot | I would set it much lower, like 10 seconds, or perhaps even 5 |
| 12:31.38 | falco_toadfoot | if there were an option for this in the config file that is |
| 12:32.01 | ManxPower | falco_toadfoot: you are going to overload your phones |
| 12:32.30 | falco_toadfoot | so it has been tried? |
| 12:33.12 | ManxPower | falco_toadfoot: qualify uses SIP OPTIONS packets and most phones give those a low priority and may not even answer the OPTIONS packet if it's busy doing something. |
| 12:33.14 | JT | falco_toadfoot: you have yet to provide a reason in favour for this |
| 12:33.34 | ManxPower | JT: if he wants to shoot himself, why should we try to stop him? |
| 12:33.49 | falco_toadfoot | JT: as I said, I need to known sooner if a device has gone offline |
| 12:33.59 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 12:34.02 | falco_toadfoot | that is, a sip peer |
| 12:34.10 | ManxPower | I don't use qualify at all |
| 12:34.17 | falco_toadfoot | I do |
| 12:34.18 | falco_toadfoot | :) |
| 12:34.49 | zapp-branigan | hi, i how can i compile the g723.1 codec for the asterisk ? |
| 12:34.56 | ManxPower | zapp-branigan: you can't |
| 12:35.35 | zapp-branigan | but the code of the codec is un the web and from itu can download the code of the g723.1 |
| 12:35.55 | ManxPower | zapp-branigan: we will not help you with that, as it is against patent law and channel policy |
| 12:36.05 | falco_toadfoot | ManxPower: sending the SIP OPTIONS more often would not reduce the time in which the device had to respond, correct? |
| 12:36.06 | zapp-branigan | ok |
| 12:36.38 | zapp-branigan | i don't know that :( |
| 12:37.03 | falco_toadfoot | ManxPower: if so, you could safely send the SIP OPTIONS at 5 second intervals |
| 12:37.12 | ManxPower | if you REALLY MUST have G723.1 then you can purchase the Digium transcoding PCI card, which should include G723.1 |
| 12:37.21 | ManxPower | falco_toadfoot: I don't know |
| 12:37.30 | ManxPower | falco_toadfoot: try it. |
| 12:37.51 | coppice | there are mercedes all over the city. i wouldn't try driving off with one, though |
| 12:38.15 | falco_toadfoot | ManxPower: will do, I'll report back |
| 12:38.52 | falco_toadfoot | ManxPower: is there an easy way to only compile the chan_sip.c so I won't have to recompile the rest? |
| 12:41.56 | ManxPower | falco_toadfoot: I don't know |
| 12:47.53 | *** join/#asterisk mharris64 (n=mharris@fedora/mharris) |
| 12:51.35 | cpm | it's a sad ole bucket-o-bolts, you don't want it. |
| 12:51.52 | coppice | they usually are. rust buckets |
| 12:52.36 | *** part/#asterisk mharris64 (n=mharris@fedora/mharris) |
| 12:52.38 | cpm | I really like the car, but much better examples exist. |
| 12:53.22 | coppice | i like no having a car. its sooooo freeing |
| 12:54.16 | cpm | no car is the best |
| 12:54.21 | cpm | (intended) |
| 12:55.29 | *** join/#asterisk bintut (n=bintut@cm246.gamma178.maxonline.com.sg) |
| 12:56.12 | coppice | you're Jeremy Clarkson, aren't you |
| 12:56.38 | cpm | I don't much like Clarkson, but he does make me laugh sometimes |
| 12:57.43 | cpm | besides, Clarkson wouldn't be caught dead in a deer dented, beat up '98 subaru outback |
| 12:59.55 | *** join/#asterisk macTijn (i=martijn@rachel.insecure.nl) |
| 13:00.01 | coppice | it might be caught dead in it if the deer impact were of the right magnitude |
| 13:02.15 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 13:02.28 | *** join/#asterisk fskrotzki_ (n=fskrot@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:02.33 | cpm | the outback does this irritating thing of clipping them off below the knees and rolling them up over the windscreen and dropping them off the side |
| 13:02.49 | cpm | like a deer scoop |
| 13:03.52 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:08.03 | mamep | how can use opal with asterisk? |
| 13:09.13 | jer | what would be the way you would suggest i set up asterisk so that i can dial an extension or some code (or some other method) to switch between an IVR and a ring-all case on a home asterisk system? that is to say, when i leave the house, i want to be able to initiate some routine that turns on the IVR, and when i get home, do the same to activate the ring-all case and turn off the IVR |
| 13:12.41 | analyysi | hi I have problem with external sip phone. It some times doesn't send register message when registration experis, is ignoreregexpire in sip.conf ok to use to workaround this problem? |
| 13:12.52 | analyysi | has anybody used ignoreregexpire parameter in sip.ocnf? |
| 13:16.26 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:20.42 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 13:22.54 | mace | hi, i'm having some difficulties with IAX connections to another server; one end claims the remote peer is OK in the 'show peers' output, but the other shows 'UNKNOWN'. also on the first host that believes it's OK, i'm getting a regular error "chan_iax2.c:5370 authenticate: Unable to find private key 'keyname' |
| 13:23.25 | mace | where keyname is the name of the key, and i've checked that as far as i can tell, the public and private key exist and are accessible on disk |
| 13:24.06 | mace | keys show however doesn't list the public or private key on that machine |
| 13:28.01 | *** join/#asterisk Darthclue (n=chatzill@zeus.nisd.net) |
| 13:28.16 | *** join/#asterisk Comete (n=Comete@krypton.saint-lo.fr) |
| 13:28.22 | Comete | hi |
| 13:31.24 | Comete | i'm using asterisk 1.2 and i would like to forward incoming calls to an external number, how can i do this ? *72 works for internal phones only... |
| 13:32.08 | *** join/#asterisk _theHub (n=_theHub@firewall.cierant.com) |
| 13:39.32 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:43.02 | tzafrir | *72 ? I think that this only works on Zaptel channels |
| 13:45.30 | *** join/#asterisk bantu (n=Miranda@p54A32C0F.dip0.t-ipconnect.de) |
| 13:46.17 | *** join/#asterisk pepo-- (n=pepOSX@190.72.146.240) |
| 13:47.24 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 13:48.34 | harpal | I have two asterisk servers running. Can I make connection netween them and than do call from one user from 1st server to second user from 2nd server? |
| 13:49.18 | Uatec | my users are really getting this hang of this SIP thing |
| 13:49.28 | Uatec | i just walked past someone in our conference room |
| 13:49.44 | Uatec | she had taken her SIP phone in with her, plugged it in in there so that she could make private calls from her own phone |
| 13:49.46 | Uatec | \o/ |
| 13:51.37 | lirakis | I want to execute an AGI application AFTER the caller hangs up. Can i use Dial() with the g parameter and then call DeadAGI() ? |
| 13:52.02 | Darthclue | How can I find out when a specific version of Asterisk was released? |
| 13:53.21 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 13:53.54 | coppice | is anyone here using audiocodes boxes? |
| 13:54.12 | Darthclue | Comete, there a couple ways of doing what you want. You'll probably have to program it into the dialplan. |
| 13:55.41 | hmmhesays | It was freaking iptables causing my intermittent no audio on answer problem |
| 13:55.50 | moemoe | telmich: ausgegruebelt? oder kam noch ne idee? |
| 13:55.52 | hmmhesays | I used to use audiocodes |
| 13:56.31 | Darthclue | hmmhesays, aren't you in ND? |
| 13:56.37 | hmmhesays | indeed |
| 13:57.49 | hmmhesays | i've never used an audiocodes box in ND though |
| 13:59.01 | hmmhesays | when I did use them they were a huge PITA |
| 14:00.11 | Darthclue | If I have a client using Asterisk 1.4.6 should I recommend an update to the latest version or just not worry about it? |
| 14:00.29 | hmmhesays | are you experiencing some issues? |
| 14:00.41 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 14:00.43 | Chris-NB | hi |
| 14:00.51 | Comete | Darthclue: i want that any user could enable or disable call forwarding for his phone, any idea ? |
| 14:00.53 | Darthclue | voice quality, but i'm not so sure they are related to this version of asterisk. |
| 14:01.00 | Chris-NB | is it possible to find out in dialplan if a particular peer has a call? |
| 14:01.12 | coppice | hmmhesays: in what way were that a PITA? |
| 14:01.55 | Comete | Darthclue: is it possible to do this with *72 / *73 ? |
| 14:01.57 | hmmhesays | coppice: I could never get the vxml scripts to work right and documentation of the web interface was poor |
| 14:02.32 | Darthclue | Comete, my first instinct would be to set a flag in some database (take your pick) that specifies where the call should go. If no entry or a null entry then send the call like normal, else forward to the specified number. |
| 14:02.39 | Comete | Darthclue: it doesn't work when i try to forward to an external number (ex: a mobile phone) |
| 14:02.52 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 14:02.54 | hmmhesays | I even had access to their official training materials. Mind you this was a couple years ago |
| 14:03.09 | Darthclue | Comete, could be a context issue, would need a pastebin of an attempt to see what is happening. |
| 14:03.24 | hmmhesays | Comete: probably because your phone is sending a 3XX redirect and asterisk is jumping into the default context |
| 14:03.35 | hmmhesays | thats usually how those things work |
| 14:03.54 | Comete | hmmhesays: what is a 3XX redirect ? |
| 14:04.07 | Comete | Darthclue: ok wait a minute |
| 14:04.38 | hmmhesays | he's going to paste what I just said I bet |
| 14:05.32 | Darthclue | so here's the deal: the client is saying that the voice (tts using cepstral and allison voice) isn't great. sounds fine to me but I probably have lower expectations. the only thing that i've seen is that some of the calls have a warning about comfort noise support. any ideas? |
| 14:05.33 | *** join/#asterisk Chris55 (n=chris@dhcp-0-30-48-92-0-88.cpe.quickclic.net) |
| 14:05.34 | hmmhesays | unless he's using freepbx |
| 14:06.09 | harpal | I have two asterisk servers running. Can I make connection netween them and than do call from user from 1st server to user from 2nd server? |
| 14:06.14 | Darthclue | if he's using any kind of gui, i can't help. i barely touch my asterisk box let alone use a gui for it. |
| 14:06.18 | Darthclue | harpal, yes |
| 14:06.25 | hmmhesays | anything starting with "Can I" is always a yes |
| 14:06.36 | *** join/#asterisk orsonork (n=orsonork@190.128.168.24) |
| 14:06.39 | orsonork | hello |
| 14:06.44 | rob0 | What about "Mother, may I?" |
| 14:06.45 | Darthclue | ehlo |
| 14:06.52 | hmmhesays | Darthclue: any transcoding involved? |
| 14:07.02 | Darthclue | rob0, the answer would then be @#$%%^ no! |
| 14:07.07 | orsonork | do you know about any free iax server? |
| 14:07.13 | orsonork | like freeworlddialup |
| 14:07.29 | rob0 | orsonork: freeworlddialup :) |
| 14:07.41 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 14:07.45 | rob0 | Also iaxtel? |
| 14:07.48 | orsonork | lol |
| 14:07.50 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 14:07.53 | orsonork | iaxtel.com doesn't work |
| 14:07.54 | orsonork | :( |
| 14:08.01 | orsonork | freeworlddialup doesn't work |
| 14:08.09 | orsonork | you can create your account |
| 14:08.12 | Darthclue | no, made sure that everything is ulaw. provider is connecting in ulaw, sound files in ulaw, call is ulaw. strangely, when i connect directly and get better sound, there is transcoding as i connect using gsm and the sound files are all ulaw. |
| 14:08.15 | rob0 | IWFM |
| 14:08.19 | orsonork | but the account isn't created on the server |
| 14:08.24 | rob0 | oh |
| 14:08.36 | orsonork | so i get registration problems with asterisk |
| 14:08.39 | rob0 | how long did you wait? |
| 14:08.53 | hmmhesays | Darthclue: I recently had iptables creating some blippy craziness on playback of audio files on a centos machine |
| 14:09.00 | rob0 | I think FWD's running on autopilot. |
| 14:09.11 | orsonork | i will keep waiting |
| 14:11.00 | Comete | Darthclue: http://pastebin.com/d72fb50b |
| 14:11.06 | Comete | hmmhesays: http://pastebin.com/d72fb50b |
| 14:11.59 | Chris55 | where are asterisk files located on FreeBSD |
| 14:12.06 | Chris55 | (the installed ones) |
| 14:12.18 | *** part/#asterisk pepo-- (n=pepOSX@190.72.146.240) |
| 14:12.49 | Comete | Chris55: are there zaptel drivers for *BSD ? |
| 14:13.13 | hmmhesays | Comete: exactly what I described |
| 14:13.33 | Comete | hmmhesays: uhm but i don't understand, what does it mean ? |
| 14:13.35 | Darthclue | Comete, it's trying to dial Local/9896043388@internal-sip. I doubt this is what you intended. Unless Local refers to a Zap channel somewhere and internal-sip is probably not the right context. |
| 14:13.38 | Chris55 | Comete: not sure im quite new to this i know it is running and was installed i just want to find the config files in order to set this up (doing some research today) |
| 14:14.06 | Darthclue | Chris55, check /etc/asterisk? |
| 14:14.25 | Comete | Darthclue: i would like it to call an external number |
| 14:14.45 | Chris55 | Darthclue: thats not the correct folder on BSD i assume... dont see it present |
| 14:14.56 | Comete | Darthclue: a mobile to be precise |
| 14:15.01 | hmmhesays | Comete: what context is your outbound Dial command in |
| 14:15.39 | hmmhesays | Comete: better yet just paste your dialplan |
| 14:16.32 | Darthclue | Chris55, where did you install asterisk at? |
| 14:17.05 | Chris55 | Darthclue: I believe it was down from the ports collection |
| 14:17.50 | hmmhesays | jump up jump up and get down |
| 14:19.04 | muiro | jump jump jump |
| 14:19.08 | muiro | everybody jump jump jump jump |
| 14:19.17 | Darthclue | Are the firewall rules still the same in 1.4? 5060 SIP and 10000-20000 RTP? |
| 14:19.26 | muiro | I can still white-rap every word of that song |
| 14:19.34 | Comete | hmmhesays: http://pastebin.com/d75956a50 |
| 14:19.45 | lirakis | hmm.. is there a way to Dial() from within an AGI app, but not leave the app if the caller hangs up? |
| 14:19.47 | muiro | yes Darthclue |
| 14:20.43 | hmmhesays | Comete: that is hardly your whole dialplan |
| 14:20.47 | coppice | I'll have a blue, blue, blue christmas...without breathing |
| 14:20.59 | _x86_ | is there a way to designate a single ZAP channel as one context for inbound and a different context for outbound? |
| 14:21.03 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:21.14 | hmmhesays | I'm probably going to crank up my peavey XXL and rock out for christmas |
| 14:21.23 | hmmhesays | maybe get myself a schecter exotic star |
| 14:21.24 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:21.46 | hmmhesays | is it me or is mantis way prettier than trac |
| 14:22.13 | tzanger | mantis sucked goat cock |
| 14:22.24 | hmmhesays | is trac better? |
| 14:22.26 | muiro | wish I could play. I had a hand injury, car accident, that ruined my rock career just like future variant 1 marty mcfly |
| 14:22.50 | hmmhesays | one that can't be rehabilitated? |
| 14:22.52 | coppice | mantis makes you want to stop developing |
| 14:23.02 | muiro | it was a complete reconstruction of three fingers |
| 14:23.07 | muiro | I cannot press frets |
| 14:23.16 | hmmhesays | I used to use it as a trouble ticket tracker at my old employer |
| 14:23.19 | muiro | by fear of shooting skeletal pain |
| 14:23.33 | hmmhesays | muiro: learn lefty |
| 14:23.39 | muiro | or piano style |
| 14:23.43 | muiro | I know |
| 14:24.03 | cpm | arg! |
| 14:24.26 | muiro | I'm very strongly right dominant |
| 14:24.28 | muiro | very |
| 14:24.37 | muiro | so it was extra effort I wasn't willing to take |
| 14:24.58 | muiro | I'll stick to reed and brass |
| 14:25.38 | hmmhesays | used soley as a trouble ticket tracker mantis worked well |
| 14:26.57 | hmmhesays | but I need something else now |
| 14:27.46 | Comete | hmmhesays: ok this the full dial plan : http://pastebin.com/d225e4420 |
| 14:28.19 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:29.52 | hmmhesays | Whats it worth to you to make this work? :D |
| 14:31.05 | hmmhesays | haha I would have anyway |
| 14:31.09 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:31.09 | hmmhesays | doesn't hurt to try |
| 14:31.15 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:32.22 | Comete | hmmhesays: when i call a forwarded number, it rings as the line was busy |
| 14:33.53 | hmmhesays | I know |
| 14:33.57 | hmmhesays | what version of asterisk? |
| 14:35.48 | harpal | I wan that If number is not availble than it dial to number 300? |
| 14:35.48 | Comete | hmmhesays: 1.2.5 |
| 14:36.23 | Qwell | Comete: That is very old. You need to upgrade. |
| 14:38.28 | hmmhesays | or keep using it if it is working |
| 14:38.34 | hmmhesays | give me a minute I have to take care of something |
| 14:38.52 | Qwell | or keep using it if you want to have known security vulnerabilities |
| 14:40.46 | Comete | hmmhesays: ok thanks |
| 14:40.51 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 14:42.23 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:42.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:46.48 | *** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net) |
| 14:47.17 | harpal | I want that if extension is not valid than it must dial 2001. so I have done exten => i,1,Dial(SIP/2001) in extension.conf |
| 14:47.32 | harpal | Is it OK? |
| 14:48.39 | *** join/#asterisk mog (i=mog@nat/digium/x-d119fb02fa56e0b2) |
| 14:48.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:49.24 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:50.21 | mamep | how can use opal with asterisk? |
| 14:50.52 | [TK]D-Fender | harpal: Only if thats part of an ivr. If you are misdialing a number from say a sip phone that jsut uses the context that's in, not it will not match |
| 14:53.18 | harpal | [TK]D-Fender, I have 2001 and 2002 two soft phones are connected. now when I try to do 101 to dial than It should dial to 2001. should it be possible? |
| 14:53.45 | [TK]D-Fender | harpal: I just told you "No". |
| 14:54.42 | harpal | [TK]D-Fender, how can I forward that request to another server? |
| 14:55.49 | harpal | I have two servers and I want that client from 1st server can make call client on 2nd server. Is it possible? |
| 14:55.55 | [TK]D-Fender | harpal: You would need to do a catch-all like "_x." or something, then check if the exten matches ANOTHER context, if so Dial Local into it, otherwise process the call another way. |
| 14:56.03 | hmmhesays | ok i'm back |
| 14:56.13 | hmmhesays | broadvoice sucks d0nk3y junk |
| 14:56.20 | [TK]D-Fender | harpal: You need to sit down and read the book and start testing. |
| 14:56.23 | [TK]D-Fender | ~book |
| 14:56.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:56.55 | [TK]D-Fender | harpal: And yoes you can set up your servers so they pass calls between each other. |
| 14:57.13 | muiro | question: should I be asking about adhearsion here or should I look for an adhearsion based channel? |
| 14:57.31 | [TK]D-Fender | harpal: This is probably in the book and is on the wiki if you search under "Asterisk dual servers". Be warned that some of the code is now deprecated. |
| 14:57.38 | De_Mon | what is adhearsion? |
| 14:57.48 | [TK]D-Fender | muiro: Not really covered here, sorry |
| 14:57.57 | muiro | alright, thanks |
| 14:58.35 | harpal | [TK]D-Fender, ok thanks a lot |
| 15:00.50 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:02.22 | hmmhesays | Comete: http://pastebin.com/m13a48075 that should do it |
| 15:02.54 | hmmhesays | can anyone recommend me a good issue tracking system? |
| 15:05.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:05.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:05.41 | Comete | hmmhesays: thanks a lot, i will try this |
| 15:08.02 | *** join/#asterisk Darthclue (n=chatzill@zeus.nisd.net) |
| 15:08.29 | file | Darthclue: you! sit in the corner. |
| 15:08.43 | Darthclue | NO! |
| 15:09.35 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.182) |
| 15:09.59 | tripps | anyone a guru at debugging php agi scripts? i'm trying to debug pbdirectory and have made the interpreter line #!/usr/bin/php -q to supress html messages but it is still spitting them out when doing agi debug |
| 15:10.55 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:11.38 | tripps | also i get "call to a member function on a non-object errors when executing $agi->verbose() functions . . . seems the $agi object hasn't been instantiated yet or something . . . weird |
| 15:14.11 | tripps | n/m - added global $agi inside function - works now . . . |
| 15:16.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:19.32 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 15:24.22 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:24.27 | tripps | is the pbdirectory agi script a base module of asterisk or is it an external module (e.g., freepbx, etc.)? |
| 15:24.28 | *** join/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 15:24.41 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 15:24.48 | agx | How i can register my digium g729 license from a web interface? it seems the "register" programs wants a console input |
| 15:24.51 | tripps | or is it part of AMP |
| 15:24.59 | [N00B] | I have been googling around all week with no success. What causes these errors: app_dial.c: Unable to forward voice or dtmf |
| 15:26.55 | tripps | seems there is a coding error in pbdirectory anyway . . . |
| 15:27.20 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:27.48 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:27.48 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:28.14 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 15:30.06 | [TK]D-Fender | tripps: ANY AGI has nothing to do with *'s coding. |
| 15:30.31 | [TK]D-Fender | tripps: All the fault of its creator/user. |
| 15:30.34 | tripps | [TK]D-Fender: roger |
| 15:30.59 | *** join/#asterisk Trionnis (i=lordkuri@s233-51-251.nap.wideopenwest.com) |
| 15:31.03 | tripps | [TK]D-Fender: seems that a script as mature as pbdirectory would be well debugged by now ;) |
| 15:31.56 | harpal | Which softphone is good for IAX/IAX2? |
| 15:31.58 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:32.03 | Trionnis | Zioper |
| 15:32.05 | Trionnis | er |
| 15:32.07 | Trionnis | Zoiper |
| 15:32.33 | Trionnis | http://www.zoiper.com/ |
| 15:32.49 | Trionnis | ~zoiper |
| 15:32.52 | mocker | Zoiper is annoying. |
| 15:32.59 | [TK]D-Fender | tripps: That script is male.... it doesn't mature, it just gets older :p |
| 15:33.00 | mocker | It comes out with updates every freaking week. |
| 15:33.04 | Trionnis | yes, but it's a decent iax phone :) |
| 15:33.11 | wwalker | I've got two servers that seem to be set up identically. both have ztdummy loaded. in one conferencing works. in the other conferencing appears to work except people can't hear each other (they can record name prompts and hear name prompts so both sides of the rtp are there |
| 15:33.36 | wwalker | zap show channels shows that meetme is grabbing a pseudo channel |
| 15:33.37 | harpal | Trionnis,ok |
| 15:33.55 | wwalker | what else should I look for to see why meetme doesn't work? |
| 15:34.55 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:34.58 | Trionnis | jbot, Zoiper is a free SIP and IAX softphone that can be found at http://www.zoiper.com |
| 15:34.58 | jbot | Trionnis: okay |
| 15:35.13 | Trionnis | ~zoiper |
| 15:35.14 | jbot | zoiper is probably a free SIP and IAX softphone that can be found at http://www.zoiper.com |
| 15:35.17 | mocker | ~xten |
| 15:35.18 | Trionnis | there we go |
| 15:35.28 | mocker | ~ |
| 15:35.30 | mocker | er.. |
| 15:35.33 | mocker | ~xlite |
| 15:35.34 | jbot | xlite is probably at download xlite at: http://snipurl.com/5tgi | and see sample configs at http://snipurl.com/5tgj, or xlite is a free SoftPhone (software phone, requires no hardware) from xten inc, |
| 15:35.53 | blitzrage | well that's not entirely true... you should use a headset :) |
| 15:35.56 | agx | avoid Zoiper+Sip i got many problems |
| 15:36.21 | Trionnis | could be user error ;) |
| 15:37.08 | mocker | Or maybe you didn't do one of the bi-weekly updates? </sarcasm> |
| 15:37.15 | Trionnis | haha |
| 15:37.34 | Trionnis | I've been using it for a while with both sip and iax, and haven't had any major problems |
| 15:37.46 | *** join/#asterisk ERROR0x0 (i=islamia@BSN-77-58-164.dsl.siol.net) |
| 15:37.47 | mocker | I tried deploying it on about 15 PCs. |
| 15:37.51 | mocker | Nothing but headaches. |
| 15:37.52 | Trionnis | the mac version has a habit of randomly exploding, but the windows one seems ok |
| 15:38.09 | mocker | xlite at least seems somewhat stable. |
| 15:38.13 | mocker | But no IAX support. |
| 15:38.58 | Trionnis | so can anyone point me at some docs about the http interface to the manager api ? |
| 15:39.16 | mocker | http interface? |
| 15:39.17 | Trionnis | the voip-info page seems to have incorrect info |
| 15:39.37 | Trionnis | http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) |
| 15:39.53 | blitzrage | sounds about right |
| 15:40.17 | Trionnis | haha |
| 15:40.19 | Trionnis | yup |
| 15:40.21 | Trionnis | "A complete list can be found by typing "show http" at the Asterisk CLI. " |
| 15:40.34 | Trionnis | show http doesn't appear to exist |
| 15:40.37 | Trionnis | heh |
| 15:40.49 | muiro | question: This is based on not knowing much about telco. Building an IVR. Extension 0 will dial to an external PSTN number. I'd prefer not to have the call processing through my asterisk system but mearly connecting the caller to the external number. Should Dial() be used for this or should I be looking up something different? |
| 15:41.02 | Trionnis | I'm trying to find a better way to generate outcalls instead of using call files or the AMI |
| 15:41.17 | [TK]D-Fender | muiro: Yes |
| 15:41.19 | blitzrage | muiro: it has to go through Asterisk if the call is in Asterisk -- you use Dial() to make a call. |
| 15:41.21 | Trionnis | need some way to get back a result if the channel is down/busy/etc |
| 15:42.02 | [TK]D-Fender | Trionnis: Both can do that. |
| 15:42.12 | Trionnis | explain? |
| 15:42.17 | muiro | [TK]D-Fender, blitzrage: thanks |
| 15:42.36 | [TK]D-Fender | Trionnis: DIALSTATUS <--- read up on your vars.... |
| 15:42.39 | ERROR0x0 | I'm new to asterisk. I have a SIP telephone from my ISP. I installed asterisk and checked with my provider about connecting it to their telephony, they said it's ok via h323. Now, what kind of an interface should i have on my asterisk box to plug the SIP line ? Is it ethernet card ? |
| 15:43.11 | Trionnis | the result has to be passed back to a separate system... I can get the info on the * box just fine, it's getting it back to the other system that's been the tricky part :) |
| 15:44.23 | [TK]D-Fender | Trionnis: How remarkably non-descript :) |
| 15:44.32 | Trionnis | haha |
| 15:45.13 | Trionnis | ok, 1 system generates the call file, scp's it over to the asterisk server, then moves it into outgoing... right now I have no easy way to pass back the dialstatus to the original server |
| 15:45.25 | *** join/#asterisk Aeudian (n=somewher@75.148.21.113) |
| 15:45.58 | Trionnis | ideally, I would use the http interface to send the call setup, then get a reply from it with the status... but it doesn't appear that I can do that without reworking parts of it |
| 15:46.27 | [TK]D-Fender | Trionnis: Of course you have to rework parts of it... it doesn't work now! :p |
| 15:46.48 | Trionnis | ... |
| 15:46.51 | Trionnis | :P |
| 15:47.20 | Trionnis | ok fine, you want to write me an app to do it? |
| 15:47.21 | Trionnis | :) |
| 15:47.34 | [TK]D-Fender | Trionnis: For everything else... there's conulting ;) |
| 15:47.40 | [TK]D-Fender | consulting* |
| 15:47.46 | Aeudian | Good Morning, we are using under voicemail.conf forcename=yes and forcegreetings=yes which allows the user to also change their password when they first enter the mailbox. The password they adjust changes in the users.conf file to whatever they make it but the password under voicemail.conf stays the same. Is this suppost to happen or did i forget to make an adjustment |
| 15:48.00 | Trionnis | yeah... but then you get these consultants that never send you a bill, and you can't pay them on time... pain in the ass, those ones |
| 15:48.04 | Trionnis | ;) |
| 15:48.24 | [TK]D-Fender | Aeudian: I didn't see words like FORCE PASS" there.... |
| 15:48.54 | [TK]D-Fender | Trionnis: Heresy! |
| 15:49.01 | Trionnis | lol |
| 15:49.33 | Aeudian | TK: i don't have force pass there, voip-info.org says if you have both forcegreeting and forcename it forces you to change your password which it does, are you saying force pass does voicemail.conf? |
| 15:50.06 | Trionnis | haha |
| 15:50.12 | Trionnis | ok, send me your resume |
| 15:50.13 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 15:50.25 | Trionnis | I know Andrew knows his shit... do you? ;) |
| 15:50.51 | Darthclue | That requires work that I'm not really motivated to do right now (The res is about 2 years out of date). |
| 15:51.18 | tzanger | I do know my shit |
| 15:51.22 | tzanger | but how did you know that |
| 15:51.30 | jameswf | I saw a guy on dirty jobs who prodused cattle feed... he knew his shit, litteraly |
| 15:51.38 | Trionnis | I'm psychic |
| 15:51.41 | Trionnis | haha |
| 15:51.52 | tzanger | if you believe in telekenesis, raise my hand |
| 15:52.01 | Trionnis | haha, that's amusing |
| 15:52.25 | Darthclue | I need to re-read the book and update my own Asterisk server as it is running some variant of 1.2. Of course, I need to get the box running again first. |
| 15:53.01 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-159385cb96a63d09) |
| 15:53.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:54.18 | blitzrage | Darthclue: that's why I just use my LinkedIn page as my resume |
| 15:54.20 | *** join/#asterisk Op3r (n=edwin@222.127.44.165) |
| 15:54.20 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 15:55.40 | Darthclue | blitzrage, I don't worry about it unless I'm actively looking for work or think I should be ready to look for work. Most consulting business comes via word of mouth or random phone calls. |
| 15:56.03 | blitzrage | indeed |
| 15:56.10 | blitzrage | I don't even think I have a copy of my resume kicking around |
| 15:56.19 | Darthclue | and I'm much happier letting other people do the work if they are capable. My 3 month old is taking up plenty of my time right now. |
| 15:56.21 | blitzrage | LinkedIn is just a lot easier to keep updated, and has all the same info |
| 15:56.50 | blitzrage | there should be a "resume export" button that builds a resume based on the info entered... that'd be neat |
| 15:57.44 | jameswf | Maybe I should make a distro called brixtox |
| 15:59.01 | Darthclue | jameswf, just call it dashit and make sure to begin a flamewar about how to pronounce it ... das hit or da shit |
| 16:00.13 | tzanger | hahaha |
| 16:00.15 | tzanger | dashit linux |
| 16:00.28 | blitzrage | dash it |
| 16:00.30 | *** join/#asterisk ming_zy1 (n=ming_zym@124.14.232.240) |
| 16:00.30 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:00.33 | tzanger | that's like the dog named askem |
| 16:00.37 | tzanger | what's his name? askem |
| 16:00.40 | blitzrage | :) |
| 16:01.23 | [TK]D-Fender | Darthclue: Its clearly Dash-it! |
| 16:01.42 | coppice | "Talk to me pussy" |
| 16:01.43 | coppice | "Me? How" |
| 16:01.52 | [TK]D-Fender | .... |
| 16:01.55 | [TK]D-Fender | *groan* |
| 16:01.57 | Darthclue | exactly |
| 16:02.07 | [TK]D-Fender | coppice: can't stand much more of this PUNishment... |
| 16:02.41 | tzanger | that is bad |
| 16:02.42 | tzanger | hahaha |
| 16:06.32 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:08.02 | tzanger | I had thanksgiving a month ago |
| 16:09.13 | [TK]D-Fender | ~idefisk |
| 16:09.14 | jbot | [~idefisk] Idefisk is now known as Zoiper, and is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 16:09.15 | [TK]D-Fender | ~zoiper |
| 16:09.16 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 16:09.17 | [TK]D-Fender | ~xlite |
| 16:09.18 | jbot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 16:09.27 | [TK]D-Fender | there, better |
| 16:09.36 | [TK]D-Fender | ~ekiga |
| 16:09.36 | jbot | [~ekiga] Ekiga (Formerly known as GnomeMeeting) is an OSS SIP soft-phone for Windows and Linux that can be downloaded from http://www.ekiga.org/ |
| 16:10.00 | tzanger | don't forget twinkle |
| 16:10.02 | tzanger | ~twinkle |
| 16:10.09 | tzanger | wow it's not there?? |
| 16:10.41 | tzanger | jbot, twinkle (http://www.twinklephone.com/) is an OSS SIP phone for KDE/Qt systems |
| 16:10.42 | jbot | that's too long, tzanger |
| 16:10.58 | tzanger | jbot, learn twinkle as (http://www.twinklephone.com/) is an OSS SIP phone for KDE/Qt systems |
| 16:10.58 | jbot | tzanger: I think you lost me on that one |
| 16:11.03 | tzanger | dammit how do I do it, [TK]D-Fender |
| 16:11.21 | [TK]D-Fender | tzanger: I'll do them up for you |
| 16:11.26 | tzanger | thanks |
| 16:11.31 | coppice | jbot, twinkle is (http://www.twinklephone.com/) is an OSS SIP phone for KDE/Qt systems |
| 16:11.31 | jbot | coppice: okay |
| 16:11.47 | tzanger | I use twinkle all the time, it's not perfect, but it's pretty good |
| 16:11.50 | tzanger | ~twinkle |
| 16:11.50 | jbot | extra, extra, read all about it, twinkle is (http://www.twinklephone.com/) is an OSS SIP phone for KDE/Qt systems |
| 16:11.55 | tzanger | yep that works |
| 16:11.57 | hmmhesays | this is freaking strange, another development in my delayed audio |
| 16:12.49 | hmmhesays | if the user sits on the phone for a couple seconds the audio then comes through |
| 16:13.01 | [TK]D-Fender | ~twinkle |
| 16:13.02 | jbot | [~twinkle] Twinkle is an OSS SIP phone for KDE/Qt systems and can be downloaded at http://www.twinklephone.com |
| 16:13.25 | [TK]D-Fender | There, standardized |
| 16:13.26 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net) |
| 16:14.50 | tzanger | danke |
| 16:15.45 | hmmhesays | i'm still getting the delayed audio with the ip 501's |
| 16:16.49 | Trionnis | so... docs on the http interface in 1.4.x? anyone? bueller? bueller? |
| 16:16.49 | hmmhesays | on on the same network segment |
| 16:16.49 | tzanger | hmmhesays: that is odd... what kind of delay? |
| 16:16.52 | hmmhesays | when the user hits the hands free button a 2-3 second delay before the audio comes through |
| 16:16.59 | tzanger | do you have a loop or bad switch causing excessive traffic? |
| 16:17.06 | tzanger | only on handsfree?? |
| 16:17.13 | hmmhesays | they only use hands free |
| 16:17.24 | tzanger | what happens if you use the handset? |
| 16:17.33 | ronr | what voip phone would you guys choose if you can choose between the polycom soundpoint IP 301 and the IP 430? |
| 16:17.37 | hmmhesays | they haven't been using the handset |
| 16:17.46 | tzanger | hmmhesays: understood, but what happens if they do? |
| 16:18.02 | hmmhesays | the problem is intermittent I don't know if it presents with the handset |
| 16:18.27 | nestAr | ronr: the 430 |
| 16:18.42 | nestAr | ronr: the 301 has no speaker phone, the 430 does IIRC |
| 16:18.47 | hmmhesays | i'm going to guess this is not a handset problem |
| 16:18.57 | ronr | nestAr: thx |
| 16:19.29 | [TK]D-Fender | ronr: Neiter. IP320/330 devalidate both other models |
| 16:19.34 | [TK]D-Fender | neither |
| 16:19.37 | [TK]D-Fender | ~softphone |
| 16:19.38 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria |
| 16:19.49 | [TK]D-Fender | Oh, and that was NOT a suggestion to use a softphone. |
| 16:19.51 | mocker | Most of the newer models of Polycoms have the silver trim. :) |
| 16:19.56 | [TK]D-Fender | was jsut testing jbot |
| 16:20.48 | coppice | mocker: very retro |
| 16:21.22 | tzafrir | [TK]D-Fender, what's this change all of a sudden? |
| 16:21.45 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 16:21.50 | [TK]D-Fender | tzafrir : what change? |
| 16:21.58 | tzafrir | of jbot's answer |
| 16:22.00 | Trionnis | he's being nice? |
| 16:22.19 | Darthclue | He's just adding fluff to his resume. Pay no attention to the man behind the curtain. |
| 16:22.37 | [TK]D-Fender | tzafrir : that was not one my MY ones to begin with and I'd rather leave the commentary for AFTER passing on useful information. |
| 16:22.41 | Trionnis | I thought it was *all* fluff... |
| 16:23.10 | [TK]D-Fender | Trionnis: this isn't "fluff", it directly usable info. |
| 16:23.13 | *** join/#asterisk asdx (n=diego@adsl-132-16.click.com.py) |
| 16:23.41 | tzafrir | [TK]D-Fender, sorry, confused you with tzanger |
| 16:23.45 | Trionnis | I was talking about your resume :) |
| 16:24.07 | [TK]D-Fender | I will be writing some new articles to summarize the common use for the various tools out there, and rebuild [av]bani's phone matrix with current info and create a better multi-point rating system. |
| 16:24.10 | tzanger | what'd I do? |
| 16:24.51 | tzafrir | you asked jbot about twinkle twice. The two answers didn't really look the same.... |
| 16:25.19 | [TK]D-Fender | While I do have a strong preference for Polycom (as do many of us) we're better served with a more comprehensive listing and grading method. |
| 16:25.24 | Darthclue | FFb3.1 is...interesting |
| 16:25.29 | tzanger | tzafrir: that's becuase tk fixed it |
| 16:25.39 | [TK]D-Fender | tzafrir :*I* did the 2nd one to match my standardized format. |
| 16:25.51 | ronr | is the Digium TE120P a good choice (given that I want a card for a E1 line)? |
| 16:26.08 | tzafrir | So it's per-factoid or a generic setting? |
| 16:26.26 | [TK]D-Fender | tzafrir : per factoid. I have redone the majority of the common ones. |
| 16:27.16 | [TK]D-Fender | tzafrir : My style is somewhat distinct, you should have no trouble identifying them. |
| 16:28.59 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 16:29.31 | *** part/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 16:29.52 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:30.21 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:31.53 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:34.54 | variable_office | does responsetimeout() not exist for 1.4? |
| 16:35.59 | outtolunc | exten => _X.,1,Set(TIMEOUT(digit)=1|TIMEOUT(response)=1) |
| 16:36.13 | nestAr | what he said |
| 16:36.16 | outtolunc | read the UPGRADE.txt file |
| 16:36.17 | blitzrage | outtolunc: that is bad format because it'll be removed in 1.6 |
| 16:36.29 | blitzrage | just an FYI |
| 16:36.32 | nestAr | Set(TIMEOUT(response)=X) |
| 16:36.32 | variable_office | do i need to set digit? |
| 16:36.39 | outtolunc | what is the format 'this week' tm <G> |
| 16:36.44 | blitzrage | outtolunc: :D |
| 16:36.48 | nestAr | Set(TIMEOUT(digit)=3) |
| 16:36.53 | blitzrage | also, you can't use pipes anymore either -- just comma's |
| 16:36.59 | nestAr | lol |
| 16:37.04 | nestAr | seriously? |
| 16:37.06 | blitzrage | yes |
| 16:37.11 | nestAr | commas to pipes back to commas |
| 16:37.16 | blitzrage | it's because it'll make escaping easier |
| 16:37.19 | outtolunc | i knew about the commas |
| 16:37.20 | nestAr | can we try semicolons next? |
| 16:37.23 | blitzrage | sure! |
| 16:37.28 | nestAr | SWEET! |
| 16:37.28 | blitzrage | patches accepted :D |
| 16:37.54 | nestAr | it's cool though, a depreciated dialplan could become valid with time |
| 16:38.12 | variable_office | nestAr, lol |
| 16:38.15 | *** join/#asterisk ai-a (n=jake2@megan.healthnet.co.uk) |
| 16:38.19 | nestAr | i should hold on to all those incarnations, and see if they'll load without warning in future versions |
| 16:38.53 | ai-a | does asterisk handle the 'message indicator' on the phone when they have a voice mail associated within voicemail.conf ? |
| 16:39.35 | Darthclue | so what's the timeline for 1.6? cause I'm running 1.2 and have no desire to upgrade but know that I need to. However, if 1.6 is around the corner then I'm gonna be lazy and just wait. |
| 16:40.15 | ronr | I'm setting up an asterisk server to replace our current gigaset isdn sets (3 ISDN BRI connections), it'll get 1 E1 with 15 ISDN lines, I need to answer the 'what if the server goes down'-question though, anyone knows if it's possible to connect a E1 to a gigaset 4170 station to create an emergency phone (with 2 lines)? |
| 16:40.28 | k0sm|k0 | hi... |
| 16:40.31 | wwalker | anyone familiar with the silence suppression/comfort noise/internal_timing issue? |
| 16:40.32 | hi365 | Darthclue: if i told you "you still have time" that would be overly optimistic |
| 16:41.34 | hi365 | what is the correct install order when using a sangoma pri? |
| 16:41.44 | Darthclue | hi365, meaning that it will be a while? or should I try running from svn? |
| 16:41.46 | variable_office | where alot of applications killed in 1.4? |
| 16:41.53 | variable_office | i know enum changed, but otherwise anything else? |
| 16:42.03 | dijungal | how do i get color in my asterisk CLI? |
| 16:42.17 | hi365 | Darthclue: my understanding is that its a lot more than a while |
| 16:42.30 | variable_office | i know enum changed, but otherwise anything else? |
| 16:42.34 | [TK]D-Fender | variable_office: please read UPGRADE.TXT , and read the rest of the docs in your source/docs folder. |
| 16:42.34 | hi365 | although the testing phase should be much shorter than 1.4 was/is |
| 16:42.39 | variable_office | oops, sorry about that double wrong window |
| 16:42.45 | Darthclue | hi365, k, guess i'll be upgrading at the same time i rebuild the hardware. |
| 16:42.53 | ai-a | Darthclue: its not a good idea to upgrade from 1.2 to 1.6 anyway ;) the reason for not upgrading to 1.4 was because of all the bugs... i imagine more bugs again in 1.6 |
| 16:43.13 | lirakis | dammit |
| 16:43.15 | nestAr | hate vitelity |
| 16:43.26 | lirakis | my Sangoma a101 T1 card went down again |
| 16:43.35 | lirakis | i dont have a clue why .. |
| 16:44.01 | Darthclue | ai-a, i haven't upgraded because I've had no need to. but the hardware crapped out last week and I'm afraid I may have to rebuild it now. |
| 16:44.18 | ai-a | Darthclue: im still using 1.2 |
| 16:44.31 | ai-a | what reason is there for upgrading ? |
| 16:45.09 | variable_office | dtmf compliance |
| 16:46.01 | Darthclue | i typically run head (if it's still called that) without issues. But I set this box up over a year ago (maybe even 2 years) and it probably needs to be brought inline with newer software that should be (in theory) better than what I am running. |
| 16:46.21 | k0sm|k0 | i would set my conf to have this: one or more phone able to call via multiple "provider"; that is: i add more provider (ex: gizmo, cheapvoip, etc), set in diaplan call to 1-xxxx-xxx placing via gizmo, 2-xxx via "another", 0-xxx and remaining via cheapnet... and finally i would set that all my phone use all these provider... but i can't specify something as context = provider-out-1, provider-out-2 |
| 16:46.38 | k0sm|k0 | anyone can tell me where i can find doc about ;D ? |
| 16:46.58 | [TK]D-Fender | k0sm|k0: ... |
| 16:46.59 | [TK]D-Fender | ~book |
| 16:47.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:47.01 | [TK]D-Fender | ^^^^ |
| 16:47.12 | k0sm|k0 | [TK]D-Fender: ah ah ah |
| 16:47.14 | k0sm|k0 | ....... |
| 16:47.21 | k0sm|k0 | ;D |
| 16:47.28 | [TK]D-Fender | k0sm|k0: This is all about DIALPLAN patterns. You can set up *'s dialplan to handle any kind of number you dial in whatever way you want. |
| 16:47.37 | [TK]D-Fender | k0sm|k0: Thats the point of Chapter 5. |
| 16:47.41 | k0sm|k0 | [TK]D-Fender: yeah |
| 16:47.41 | k0sm|k0 | ;D |
| 16:47.45 | k0sm|k0 | just wget ;D |
| 16:47.57 | [TK]D-Fender | k0sm|k0: This is something you're jsut going to have to suit down and read and start playing with. |
| 16:49.29 | [TK]D-Fender | k0sm|k0: Small sample for you though : exten => _5x.,1,Dial(SIP/out-cheapnet/${EXTEN:1}) <- theoretical sample for taking any number starting with a 5, stripping the 5 and dialing it out a specific peer pointed to a provider (which you of course would ahve to have set up in sip.conf |
| 16:50.35 | k0sm|k0 | yes, this is yet working |
| 16:51.02 | k0sm|k0 | now, all call via cheapnet (my default) |
| 16:51.08 | hi365 | what is the install order whn using a sangoma pri card? |
| 16:51.19 | k0sm|k0 | i can't able to call via gizmo, i.e.... 1-x... |
| 16:51.26 | k0sm|k0 | i paste something |
| 16:52.00 | [TK]D-Fender | k0sm|k0: exten => _1x.,1,Dial(SIP/out-gizmo/${EXTEN:1}) |
| 16:52.13 | [TK]D-Fender | k0sm|k0: Again, jsut a sample, nothing you are likely to be able to use as-is |
| 16:52.53 | [TK]D-Fender | k0sm|k0: First you need to figure out what the problem is : dialplan , channel config (sip/iax/etc), networking, etc |
| 16:52.58 | [TK]D-Fender | ~pb |
| 16:52.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:52.59 | k0sm|k0 | uh... i understand... it sound like iptables... i have to specify any rules |
| 16:53.23 | [TK]D-Fender | hi365: libpri, zaptel, wanpipe (will rebuild Zaptel), asterisk |
| 16:53.24 | k0sm|k0 | [TK]D-Fender: i had specify with include.. but it seems not working |
| 16:53.35 | k0sm|k0 | [to-gizmo] |
| 16:53.35 | k0sm|k0 | exten => _1747.,1,SetCallerID("Your Name" <Your Num>) |
| 16:53.35 | k0sm|k0 | exten => _1747.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r) |
| 16:53.35 | hi365 | [TK]D-Fender thanks |
| 16:53.37 | [TK]D-Fender | k0sm|k0: pastebin it and we'll take a look |
| 16:53.50 | k0sm|k0 | and after: include => [to-gizmo] |
| 16:53.52 | k0sm|k0 | i have this |
| 16:54.13 | [TK]D-Fender | koszik: no [] around the context name. like this : include => to-gizmo |
| 16:55.07 | [TK]D-Fender | k0sm|k0: And warning that you are using SetCallerID which was deprecated in 1.2. Make sure your code is using the latest form that your version supports.... |
| 16:56.42 | k0sm|k0 | warning don't appear.. however i delete |
| 17:00.25 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 17:05.56 | *** join/#asterisk PepOSX (n=pepOSX@190.72.146.240) |
| 17:07.13 | variable_office | in sip.conf if i have nat=yes i need to set canreinvite=nonat in order for reinvites to be turned off? |
| 17:07.20 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.104.86) |
| 17:07.23 | variable_office | canreinvite=no will not work? |
| 17:07.41 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 17:08.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:08.29 | *** join/#asterisk b_d (n=brian@209.240.42.151) |
| 17:15.30 | [TK]D-Fender | variable_office: Set "canreinvite=no" EVERYWHERE. |
| 17:15.37 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:16.38 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 17:18.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:20.11 | variable_office | [TK]D-Fender, in the UPDATES.txt thing though it says to use canreinvite=nonat , or is that invalid? |
| 17:20.35 | [TK]D-Fender | variable_office: taht allows for creinvites where possible between respective ends of a given call |
| 17:21.27 | Uatec | hey, for some reaosn, i'm getting: Nov 20 17:20:35 ERROR[18516]: chan_sip.c:11071 handle_request_subscribe: Got SUBSCRIBE for extension 763@internal from 192.168.0.4, but there is no hint for that extension in my CLI |
| 17:21.34 | Uatec | i don't konw why it's happening |
| 17:21.41 | Uatec | or where it's coming from |
| 17:21.49 | Uatec | 192.168.0.4 isn't even on our subnet |
| 17:22.05 | variable_office | [TK]D-Fender, ah, so canreinvite=no is guaranteed whether nat is yes or no; to make sure nothing reinvites |
| 17:22.09 | Uatec | where could it be coming from? |
| 17:22.12 | variable_office | ? |
| 17:22.37 | [TK]D-Fender | variable_office: Correct |
| 17:22.58 | [TK]D-Fender | Uatec: a remote phone that is behind NAT and not set up correctly. |
| 17:23.31 | [TK]D-Fender | Any Ops around? |
| 17:23.36 | Qwell | ? |
| 17:24.11 | hi365 | [TK]D-Fender: i rebuilt everything (libpri, zaptel, wanpipe, asterisk, asterisk addons) in thata order nut when i log in to asterisk -r there are no zap/pri applications available. what can i check? zaptel/zapata seem to be correct... |
| 17:24.20 | *** join/#asterisk marcmini (n=mini@71-222-126-87.ptld.qwest.net) |
| 17:24.25 | hi365 | nut=but |
| 17:24.59 | marcmini | I tried installing Zapata, but on boot it says module not found also there is no /dev/zap do I need to compile my kernel? |
| 17:25.46 | [TK]D-Fender | hi365: go verify that wanrouter is running normal, pastebin yrou configs, test with ztcfg -vvvv and verify that can_zap.so is present and loaded. attempt to manually load if it is not and see what happess. |
| 17:25.50 | [TK]D-Fender | happens |
| 17:26.00 | hi365 | cool. will do |
| 17:28.12 | tzafrir | marcmini, Zaptel, right? |
| 17:28.24 | tzafrir | What version? On what distro? |
| 17:28.58 | marcmini | centos 5 |
| 17:29.03 | Qwell | ~wifi |
| 17:29.04 | jbot | rumour has it, wifi is see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 17:29.04 | tzafrir | hi365, if there are no zap apps available, chan_zap has probably failed to load |
| 17:29.08 | tzafrir | look at the logs |
| 17:29.17 | hi365 | tzafrir: kanireh |
| 17:29.28 | hi365 | but why?? |
| 17:29.40 | tzafrir | hi365, look as /var/log/asterisk/{messages,full} |
| 17:29.51 | hi365 | k |
| 17:30.09 | tzafrir | marcmini, what version of Zaptel? |
| 17:30.12 | marcmini | 1.4.6 |
| 17:30.37 | tzafrir | What is the output of: zaptel_hardware |
| 17:30.51 | marcmini | I think it has to do with me not being able to decipher : "You need to recompile your kernel with CONFIG_CRC_CCITT=y or CONFIG_CRC_CCITT=m. To do this, run 'make menuconfig' and change it under 'Library Routines -> CRC-CCIT functions'. If you compiled it as a module, load it, otherwise reboot." Dont know though |
| 17:31.19 | marcmini | none |
| 17:31.29 | marcmini | zaptel_hardware = no output |
| 17:31.40 | tzafrir | marcmini, if you don't use a custom kernel, don't worry about those |
| 17:31.50 | marcmini | ok |
| 17:32.02 | marcmini | on boot I do see module not found for Zapata |
| 17:32.16 | marcmini | rather zaptel maybe |
| 17:32.33 | tzafrir | modinfo zaptel - returns an output at all? or just an error? |
| 17:33.01 | dijungal | how do i get colors in my CLI? |
| 17:33.05 | marcmini | modinfo: could not find module zaptel |
| 17:33.14 | tzafrir | marcmini, you have no zaptel hardware, just need ztdummy for timing, right? |
| 17:33.15 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 17:33.16 | marcmini | could it be that I used the wrong kernel sources |
| 17:33.20 | marcmini | thats right |
| 17:33.25 | marcmini | no zaptel hardware |
| 17:33.38 | tzafrir | marcmini, what is the output of: ./install_prereq test |
| 17:33.54 | tzafrir | sorry: |
| 17:34.02 | marcmini | Kernel source not found. Installing kernel-devel-2.6.18-8.el5 |
| 17:34.04 | tzafrir | marcmini, what is the output of: ./install_prereq test | grep -v '^#' |
| 17:34.55 | tzafrir | try running the command from there: |
| 17:35.04 | tzafrir | ./install_prereq install |
| 17:35.09 | marcmini | installing those packages |
| 17:35.40 | marcmini | so I should probably re compile asterisk after zapata? |
| 17:36.11 | marcmini | <PROTECTED> |
| 17:36.15 | marcmini | error |
| 17:38.59 | marcmini | I removed kernel-devel-2.6.18-8.1.15.el5 and installed kernel-devel-2.6.18-8.el5 rebooting now |
| 17:39.51 | hi365 | tzafrir: [Nov 21 01:29:42] ERROR[3791] chan_zap.c: Unknown signalling method 'pri_cpe' |
| 17:39.51 | hi365 | [Nov 21 01:29:42] ERROR[3791] chan_zap.c: Signalling must be specified before any channels are |
| 17:40.06 | tzafrir | well, centos's handling os kernel headers is less than stellar :-( |
| 17:40.25 | tzafrir | hi365, you built asterisk with no support for libpri |
| 17:40.44 | hi365 | is there on option in ./configure for that?? |
| 17:40.55 | tzafrir | you may need to re-run ./configure again after installing libpri |
| 17:41.13 | hi365 | ill try |
| 17:43.48 | marcmini | installing kernel and kernel-devel packages hopefully that will help |
| 17:45.38 | tzafrir | hmm... you installed "kernel"? what version, exactly? |
| 17:45.43 | tzafrir | rpm -q kernel |
| 17:46.06 | marcmini | not sure, rebooting now |
| 17:47.57 | marcmini | test completed successfully |
| 17:48.52 | marcmini | rpm -q kernel = kernel-2.6.18-8.el5 <br> kernel-2.6.18-8.1.15.el5 |
| 17:50.36 | marcmini | what does : need ztdummy for timing, mean? |
| 17:51.12 | marcmini | Loading zaptel framework: [ OK ] |
| 17:51.24 | marcmini | Running ztcfg: [ OK ] |
| 17:54.15 | [TK]D-Fender | marcmini: He's asking if you plan on using MeetMe / IAX trunking. |
| 17:55.13 | hi365 | [TK]D-Fender, tzafrir Thanks a lot! |
| 17:55.37 | [TK]D-Fender | hi365: You're welcome. |
| 17:55.39 | marcmini | this is my first real compile / install with zaptel so I am not sure |
| 17:56.46 | *** join/#asterisk awannabe (n=brad@ip24-251-152-67.ph.ph.cox.net) |
| 17:57.11 | awannabe | hi guys, i got a pair of boxes setup with IAX trunks, and i cant pass calls to them, i keep getting number@default is invalid, i have context= in both sides in iax.conf why is this? |
| 17:57.50 | bkruse | awannabe: are you sure that they have authority? aka they are getting match to an entry in iax.conf? |
| 17:58.04 | awannabe | i think its my dial command that is bad |
| 17:58.07 | bkruse | because the fact its number@default is because possibly its not getting match |
| 17:58.21 | awannabe | Dial(IAX2/iax-peer-name/123456,200) |
| 17:58.30 | bkruse | dial(iax2/user_entry_in_users.conf/12345) |
| 17:58.32 | bkruse | that should work fine. |
| 17:58.49 | bkruse | are you using registration or static hosts? |
| 17:59.11 | awannabe | staic hosts |
| 17:59.29 | bkruse | perfect |
| 17:59.30 | awannabe | both are showing connected and all that, peer looks good |
| 17:59.32 | bkruse | pastebin both configs please |
| 17:59.35 | bkruse | right |
| 17:59.55 | bkruse | my deal is, the peer probably is NOT getting matched, so think of the asterisk box receiving the call as seeing it as an unauthenticated call |
| 18:00.27 | *** join/#asterisk mmdd (n=mmdd@142.165.159.222) |
| 18:00.41 | mmdd | hello |
| 18:01.30 | awannabe | nowhttp://pastebin.com/m505fea8b |
| 18:01.50 | awannabe | im tying to pass calls from customerA to core |
| 18:01.56 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 18:03.52 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:04.29 | awannabe | bkruse, i have the context in each peer set to the context on that same local box, which I thought was correct |
| 18:07.25 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:08.23 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 18:11.09 | *** join/#asterisk icel (n=icel@63.78.162.5) |
| 18:12.13 | icel | hey, if I just want to make a basic backup of an asterisk server, what directories should I copy (I'd like to just be able to untar them on an identical machine and have it 'work') |
| 18:12.38 | icel | do i need anything besides /etc/asterisk? |
| 18:12.48 | waKKu | binaries ? |
| 18:13.40 | icel | If i had it installed on another machine already |
| 18:13.43 | waKKu | do a make install DEST=/somedir ; add your configs from /etc/asterisk and pack /somedir... I think this can wok |
| 18:13.49 | waKKu | oh :) |
| 18:14.35 | waKKu | so if u has only asterisk (no hylafax, spandsp ...) you just need contents from /etc/asterisk |
| 18:14.39 | icel | Do I need to copy anything in the /var/lib/asterisk or /var/spool/asterisk dir or anything? |
| 18:14.44 | icel | ok |
| 18:14.46 | icel | thx |
| 18:15.24 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 18:23.21 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:35.17 | *** join/#asterisk myiagy (n=myiagy@201-14-162-236.fnsce701.dsl.brasiltelecom.net.br) |
| 18:36.39 | deever | hello! what does this mean? |
| 18:36.42 | deever | chan_sip.c:7364 sip_reg_timeout: -- Registration for 'NUMBER@sip.backbone.ch' timed out, trying again (Attempt #1) |
| 18:37.25 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 18:37.35 | Greek-Boy | interesting in 1.4 is verbose limit |
| 18:37.37 | Greek-Boy | lg-asterisk01*CLI> core set verbose 9999999999999 |
| 18:37.37 | Greek-Boy | Verbosity is at least 2147483647 |
| 18:37.39 | Greek-Boy | very high :P |
| 18:37.49 | deever | i can do outbound calls over this provider, but i don't receive incoming calls |
| 18:39.08 | deever | and i've found out (with tcpdump) that the sip.backbone.ch answers with status 500, "could not create transaction" |
| 18:39.20 | deever | s/the// |
| 18:39.46 | nestAr | deever: means that they are not letting you register |
| 18:40.07 | nestAr | have you checked with your provider to see why they are hating on you? |
| 18:40.25 | deever | but with wengophone (directly), it works! ;) |
| 18:40.43 | deever | nestAr: no, not yet |
| 18:41.12 | deever | and outgoing calls work with asterisk! |
| 18:41.17 | nestAr | are you specifying the info it expects to get? |
| 18:41.27 | deever | strange, isn't it? |
| 18:41.29 | nestAr | outgoing valls typically use a different set of info |
| 18:41.55 | nestAr | for my provider, i actually have two seperate sip peers, one for inbound one for outbound |
| 18:42.26 | nestAr | you also don't need to register for outbound calls |
| 18:42.28 | deever | well, i tried many configurations, but neither one did work |
| 18:42.40 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:42.46 | nestAr | so that's why it works despite the registeration thing. |
| 18:42.58 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:42.59 | deever | m-hm |
| 18:44.23 | Greek-Boy | has anybody here succeeded with speech recogniztion in asterisk? |
| 18:46.04 | *** join/#asterisk gardo (n=gardo@121.97.195.160) |
| 18:48.18 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 18:49.33 | Navion | What's the best way to save all the asterisk profiles to do a clean reinstall? |
| 18:52.20 | deever | nestAr: which keyword do you have in your in- and outbound sections? |
| 18:52.32 | bkruse | Navion: tar -czvf asterisk.config.backup.`date +%s`.tar.gz /etc/asterisk |
| 18:52.47 | bkruse | for your CONFIGS ONLY, that does not include modules (which if you update, could break.) |
| 18:54.00 | Navion | OK, I think my install was corrupted. A bunch of missing stuff. I've done a lot of work to get things somewhat working but now I'm stuck and need a clean install. |
| 18:56.00 | nestAr | deever: hang on |
| 18:57.41 | *** join/#asterisk Op3r (n=edwin@203.177.233.39) |
| 18:58.40 | Navion | What is "asterisk.config.backup? find doesn't find it. |
| 18:59.05 | deever | nestAr: ? |
| 18:59.09 | nestAr | deever: http://www.pastebin.ca/791084 |
| 18:59.57 | deever | nestAr: oh thanks, i'll try it out! ;) |
| 19:00.05 | sigmounte | anyone can help me with ztdummy ? |
| 19:00.33 | Navion | Nevermind... (dough!) |
| 19:03.32 | tzafrir | I'm trying to use RxFAX and TxFAX and so far can't get even a simple fax from TxFax from one to the other |
| 19:03.59 | tzafrir | I put a simple tiff file at /tmp/test.tiff |
| 19:05.25 | tzafrir | dialplan show tests shows, among others: |
| 19:05.29 | tzafrir | <PROTECTED> |
| 19:05.30 | tzafrir | <PROTECTED> |
| 19:05.37 | tzafrir | I run: |
| 19:05.54 | tzafrir | originate Local/faxsend@tests extension fax@tests |
| 19:06.38 | tzafrir | I get basically two channels running. No sign of even a handshake or even debug messages from RxFax |
| 19:06.47 | tzafrir | No immediate hangup, though |
| 19:09.01 | *** join/#asterisk NeonLevel (n=NeonLeve@189.169.139.125) |
| 19:09.07 | *** join/#asterisk grEvenX (n=even@84.52.254.130) |
| 19:15.16 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 19:15.22 | msetim | hi guys |
| 19:15.32 | blitzrage | ohio |
| 19:16.32 | tzafrir | sigmounte, I need to go now. But it would basically help if you'd fill in some details |
| 19:16.43 | tzafrir | your system, what works, what doesn't |
| 19:16.53 | tzafrir | What distro do you use? What version of Zaptel? |
| 19:17.43 | msetim | I have a question. Asterisk make locks When I perform some reload ( e.g sip reload )? |
| 19:20.33 | *** join/#asterisk Darkstorm^ (n=DarkStor@chello062178240242.3.14.wu-wien.teleweb.at) |
| 19:20.34 | Darkstorm^ | hi@ |
| 19:20.37 | *** part/#asterisk myiagy (n=myiagy@201-14-162-236.fnsce701.dsl.brasiltelecom.net.br) |
| 19:21.33 | [hC] | so this is interesting. Paging to 50-60 phones is way quicker/more reliable when using ztdummy over zaptel+wanpipe (sangoma) |
| 19:21.48 | [hC] | when using the sangoma, people got 3-4 second delays before all the phones joined the meetme |
| 19:21.52 | [hC] | with ztdummy, it was instant. |
| 19:23.06 | Darkstorm^ | any idea why all incoming calls do just nothing? i'm calling and the call is waiting for response - asterisk notes the call - the incoming rout is set to "any DID / any CID" but nothing happen |
| 19:23.07 | Darkstorm^ | *happens |
| 19:23.39 | Darkstorm^ | regardless of which destination i'm selecting for the incoming route |
| 19:24.06 | *** join/#asterisk Op3r (n=edwin@203.177.233.72) |
| 19:25.07 | *** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket) |
| 19:26.53 | JayTee52 | I just recently added a TDM400P card to our Asterisk server. I added it to the first PCI slot on the bus and when I restart the server the TDM400P is assigned channels 1-4 and the X100 cards (which will be replaced shortly with a second TDM400) grab channels 5 and 6. I performed a soft reboot of the server today and one of the X100 cards grabbed channel 1 and the TDM400 then assigned channels 2 through 5. Is there a way to force th |
| 19:27.37 | hmmhesays | [hc] that seems odd |
| 19:27.42 | hmmhesays | what version of asterisk and zaptel? |
| 19:27.54 | JayTee52 | 1.2.11 |
| 19:28.36 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:28.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:28.37 | [hC] | hmmhesays: ive seen it a few places actually. this particular install is ast 1.2.17 and zap 1.2.16 |
| 19:29.01 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:29.02 | Darkstorm^ | hi russellb |
| 19:29.35 | russellb | hello |
| 19:30.32 | Darkstorm^ | i know i'm reasking the same question again but i'm getting mad - solved 2 problems with the c450ip an the outgoing calls today but i'm not able to solve this prob: any idea why all incoming calls do just nothing? i'm calling and the call is waiting for response - asterisk notes the call - the incoming rout is set to "any DID / any CID" but nothing happen |
| 19:32.11 | Darthclue | Darkstorm^, pastebin some cli output and config info and maybe someone will be able to assist... |
| 19:32.45 | Darkstorm^ | Darthclue, would be an idea but i think noone is interested in such a nooby problem, but I'll nopaste some stuff ;) |
| 19:33.31 | *** join/#asterisk mmdd (n=mmdd@142.165.159.222) |
| 19:33.34 | Darthclue | none of us find problems interesting, we prefer to be innovative and bleeding edge...no really, we do. |
| 19:33.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 19:34.02 | hmmhesays | [hC]: i'm having an intermittent problem where there is no audio on the answer sip phone when the call comes from a sangoma a200 |
| 19:34.21 | *** join/#asterisk mmdd (n=mmdd@142.165.159.222) |
| 19:35.42 | [hC] | hmmhesays: i have that too sometimes. only recently. |
| 19:35.50 | [hC] | hmmhesays: people claim they answer a call and only get half the audio. |
| 19:36.21 | [hC] | hmmhesays: the only thing in particular for me is that it happens when their internet connectivity is gone (and thus failing over to analog) but i dont know if thats the cause of the one way audio or not. |
| 19:39.17 | hmmhesays | mine is simple a200->asterisk->ip501 |
| 19:41.33 | *** join/#asterisk darkstorm^ (n=DarkStor@chello062178240242.3.14.wu-wien.teleweb.at) |
| 19:41.36 | darkstorm^ | nice |
| 19:41.38 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 19:41.39 | Darthclue | and it only happens on the a200? what if you have calls only on the sip side or only on the a200 side? |
| 19:41.44 | darkstorm^ | after a failed call my router reboots -.- |
| 19:42.17 | sigmounte | tzafrir, debian , custom kernel , not zaptel hardware , asterisk 1.4.2 , trying to use meetme , but without ztdummy it doesn't work at all (normal) and with zt dummy i have no sound and "rtc: lost some interrupts at 1024hz" filling my log |
| 19:43.13 | tzafrir | sigmounte, what kernel exactly? |
| 19:43.22 | sigmounte | 2.6.18 custom |
| 19:45.22 | tzafrir | sigmounte, anyway, 1.4.2 is old by now. If this is a new install, do use something newer |
| 19:46.16 | sigmounte | yep , but the problem is from ztdummy (got the problem when modprobing it ) and ztdummy came from zaptel tgz (1.4.6) the latest i've found |
| 19:47.16 | tzafrir | Right |
| 19:47.40 | tzafrir | Heard of this problem. Nott really sure how to fix it |
| 19:47.51 | sigmounte | my life :) |
| 19:48.02 | tzafrir | hmm... do you happen to have HZ=1000 in your custom kernel? |
| 19:48.09 | sigmounte | yes |
| 19:48.19 | sigmounte | let try a cat .config | grep 1000 to be sure |
| 19:48.22 | *** join/#asterisk lirakis (n=user@69.176.204.146) |
| 19:48.40 | lirakis | woo new asus eee pc |
| 19:48.43 | tzafrir | try editing ztdummy and disable USE_RTC |
| 19:48.50 | tzafrir | ztdummy.c , that is |
| 19:48.55 | sigmounte | ok thanks |
| 19:49.04 | tzafrir | and then re-run 'make modules' |
| 19:49.05 | sigmounte | but oups # CONFIG_HZ_1000 is not set |
| 19:49.05 | sigmounte | CONFIG_HZ=250 |
| 19:49.11 | tzafrir | bah |
| 19:49.12 | sigmounte | i feel stupid |
| 19:49.19 | sigmounte | grrrr |
| 19:49.20 | lirakis | still dont know why my sangoma a101 has died twice, randomly |
| 19:49.30 | darkstorm^ | anyone any idea in which file the inbound is stored if i'm using freepbx to configure? |
| 19:49.41 | Qwell | ~freepbx |
| 19:49.41 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:49.51 | darkstorm^ | thx |
| 19:50.31 | sigmounte | thanks tzafrir |
| 19:50.44 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 19:51.11 | tzafrir | sigmounte, sorry. IIRC you can twak rtc a bit with some sysfs settings. |
| 19:51.15 | Kobaz | okay so i have a dialin problem... http://pastebin.ca/791182 |
| 19:51.17 | tzafrir | But I never really tried |
| 19:51.56 | Kobaz | asterisk is registered with voicepulse successfully, according to the iax2 debug, asterisk recieves the call, but it never hits anything in the dialplan... i have no idea what's going on |
| 19:52.12 | sigmounte | wow it work !! |
| 19:52.26 | sigmounte | tzafrir, thanks thanks thanks the commented USE-RTC work fine ! |
| 19:52.38 | lirakis | Kobaz: are you sure voicepulse is sending the 1? |
| 19:52.49 | lirakis | Kobaz: try a more general pattern for trouble shooting |
| 19:52.50 | Kobaz | yeah it's right there in the iax debug |
| 19:53.03 | tzafrir | sigmounte, but it will not do anything good unless you have HZ=1000 |
| 19:53.11 | sigmounte | yep |
| 19:53.15 | Kobaz | slightly monkeied to protect the innocent, but that's the pattern |
| 19:53.22 | sigmounte | but kernel recompiling for later tonigh :) |
| 19:53.29 | tzafrir | It will give you wayyy wrong timing |
| 19:53.40 | lirakis | Kobaz: and what does cli show? ... do you see anything regarding the calls routing as it comes in |
| 19:53.44 | Kobaz | nothing |
| 19:54.05 | Kobaz | lirakis: if i dial 17148761466 as a local extension it works fine |
| 19:54.11 | Kobaz | er |
| 19:54.25 | Kobaz | the number that's in there |
| 19:54.31 | Kobaz | like 17013334444 |
| 19:54.35 | Kobaz | if i load up a soft phone |
| 19:54.40 | Kobaz | and dial that, it works fine |
| 19:54.54 | Kobaz | if i call from an outside line, and try to get a dialin to asterisk from voicepulse |
| 19:55.01 | Kobaz | all i get are iax2 packets |
| 19:55.06 | Kobaz | and no call routing |
| 19:56.16 | [TK]D-Fender | Kobaz: PASTEBIN is your friend... |
| 19:56.17 | hmmhesays | http://www.pastebin.ca/791196 <-- is this something normal in dmesg? |
| 19:56.17 | [TK]D-Fender | ~pb |
| 19:56.18 | jbot | [~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 |
| 19:56.25 | Kobaz | http://pastebin.ca/791182 |
| 19:56.27 | Kobaz | you mean that? |
| 19:56.37 | Kobaz | that i already pasted :P |
| 19:57.06 | lirakis | Kobaz: .. whats the local context? |
| 19:57.39 | Kobaz | i'll add that in |
| 19:57.46 | Kobaz | but that's for dialing out |
| 19:57.49 | Kobaz | which works |
| 19:58.08 | lirakis | Kobaz: its possible your call isnt routing to the context you believe it is... so try and debug that |
| 19:58.17 | Kobaz | http://pastebin.ca/791198 |
| 19:58.22 | Kobaz | that's with the local context |
| 19:58.39 | Kobaz | outgoing works fine, incoming is borked |
| 19:58.49 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 19:58.55 | Kobaz | lirakis: how would i see what context it's trying to use |
| 19:58.58 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 19:59.05 | lirakis | Kobaz: because in [general] in iax.conf you define context=local ... so any thing that doesnt match would fall to 'local' |
| 19:59.34 | Kobaz | yeah |
| 19:59.56 | lirakis | Kobaz: watch cli .. use NOOP("im here") to see where the call is hitting your dialplan .. obviously replace im here with some thing more usefull |
| 20:00.35 | Kobaz | yeah, k |
| 20:00.46 | [TK]D-Fender | Kobaz: Follow their guide... http://connect.voicepulse.com/samples/iax.sample |
| 20:00.48 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 20:01.05 | [TK]D-Fender | Kobaz: back link ... http://voicepulse.custhelp.com/cgi-bin/voicepulse.cfg/php/enduser/std_adp.php?p_faqid=67&p_created=1097514602&p_sid=9IkQZdRi&p_accessibility=0&p_lva=&p_sp=cF9zcmNoPTEmcF9zb3J0X2J5PSZwX2dyaWRzb3J0PSZwX3Jvd19jbnQ9MTcmcF9wcm9kcz0yJnBfY2F0cz0wJnBfcHY9MS4yJnBfY3Y9JnBfcGFnZT0xJnBfc2VhcmNoX3RleHQ9QXN0ZXJpc2s*&p_li=&p_topview=1 |
| 20:01.13 | Kobaz | that's a long one |
| 20:01.46 | *** join/#asterisk cesar_CR (n=cesar@201.195.35.62) |
| 20:05.12 | hmmhesays | http://www.pastebin.ca/791196 |
| 20:05.26 | Kobaz | yeah |
| 20:05.27 | Kobaz | umm |
| 20:05.37 | Kobaz | it doesn't look too bad, they don't look like errors |
| 20:10.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 20:14.58 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.37.23) |
| 20:15.38 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 20:18.14 | Kobaz | so |
| 20:18.31 | Kobaz | i put some noops in local and abc |
| 20:18.36 | Kobaz | nothing is getting hit |
| 20:18.50 | Kobaz | if i do a dial in the console, they run fine |
| 20:18.53 | [TK]D-Fender | Kobaz: can you give a complete CLI dump with iax2 debug enabled. |
| 20:19.01 | Kobaz | yeah it's in the pastebin, sec |
| 20:19.09 | Kobaz | http://pastebin.ca/791198 |
| 20:19.16 | Kobaz | that's the entire thing |
| 20:19.37 | Kobaz | that's everything in the console after placing the call, along with verbosity 6 |
| 20:19.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:20.51 | Kobaz | here's the new version |
| 20:20.57 | Kobaz | http://pastebin.ca/791253 |
| 20:21.15 | [TK]D-Fender | Kobaz: allow=ulaw,g726,gsm,ilbc,alaw,adpcm<- last I checked you couldn't do those on one row |
| 20:21.34 | Kobaz | it works on my other setups |
| 20:21.40 | Kobaz | hmm |
| 20:22.29 | Kobaz | okay, i split them onto multiple lines i'll see if it makes any difference |
| 20:22.48 | [TK]D-Fender | kob and "called number" is 17013334444 and nothing in [local] matches. try to FORCE a match tos ee where its landing. |
| 20:22.51 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 20:23.57 | Kobaz | exten => _1NXXXXXXXXX,1,NOOP(local) |
| 20:23.58 | Kobaz | that matches |
| 20:24.08 | Kobaz | if i do a forced match... it doesn't do anything either |
| 20:25.10 | [TK]D-Fender | Kobaz: that line wasn't in the pastebin you jsut gave me. |
| 20:25.39 | Kobaz | 15:20 <Kobaz> here's the new version |
| 20:25.39 | Kobaz | 15:21 <Kobaz> http://pastebin.ca/791253 |
| 20:25.46 | Kobaz | i pasted a new one with the updated stuff |
| 20:26.06 | Kobaz | something is very wrong somewhere |
| 20:26.10 | Kobaz | there's no other debugging i can turn on? |
| 20:29.10 | Kobaz | according to the console, asterisk isn't doing anything after it gets the initial iax packets |
| 20:29.16 | *** join/#asterisk lclimber (i=lcanelon@200.82.188.26) |
| 20:29.47 | [TK]D-Fender | Kobaz: Please make your config match their sample.... http://connect.voicepulse.com/samples/iax.sample |
| 20:31.24 | lclimber | hello everyone, and thanks in advance for your help, i work on a 2000 users company, we want to install an asterisk server to comunicate all users within, we have 10 outgoing lines which the users will use to make calls outside the company, which means 100 simultaneus calls, what kind of hardware do i need as the asterisk box, and how many boxes would you recomend |
| 20:32.18 | Kobaz | define "10 outgoing lines" |
| 20:32.30 | lclimber | yes |
| 20:32.39 | Darthclue | 10 != 100? |
| 20:32.40 | lclimber | that is 10 externel lines |
| 20:33.07 | fujin_ | how do you proposte to get 100 simultaneous calls on 10 lines? |
| 20:33.14 | Kobaz | that's what i was wondering |
| 20:33.14 | fujin_ | MOOOLTIPLEXING? |
| 20:33.15 | lclimber | 100 simultaneus calls becouse we hace so many users |
| 20:33.32 | Kobaz | you need 100 outgoing lines to support 100 concurrent users |
| 20:33.50 | Darthclue | so you mean that you want to support up to 100 calls over the entire system but you have a total of 10 external lines? |
| 20:33.57 | [TK]D-Fender | Kobaz: fujin_ think that MAYBE he's including INTERAL calls in that figure... |
| 20:34.06 | lclimber | yes |
| 20:34.17 | Kobaz | that makes more sense then |
| 20:34.23 | fujin_ | damn, so no multiplexing? |
| 20:34.24 | fujin_ | o_0 |
| 20:34.29 | [TK]D-Fender | His wording is rough, spare a few seconds to think on this... |
| 20:34.34 | Darthclue | are these 10 analog lines? |
| 20:34.35 | fujin_ | NEVERRR |
| 20:34.43 | lclimber | i mean, we have 2000 users, those users will try to make external calls |
| 20:34.51 | lclimber | yes |
| 20:35.02 | [TK]D-Fender | lclimber: You sure 10 lines are enough for that many users? |
| 20:35.27 | fujin_ | wow, 2000 users, and only 10 simultaenous external calls? |
| 20:35.31 | fujin_ | that's pretty impossible. |
| 20:35.41 | fujin_ | are you going to roll 2000 hardphones? |
| 20:35.51 | [TK]D-Fender | fujin_: Not impossible, just unusual |
| 20:36.03 | fujin_ | unusually stupid? |
| 20:36.15 | riddlebox | wohoo my boss just told me he quoted an asterisk system to a customer! |
| 20:36.26 | fujin_ | WOHOO |
| 20:36.42 | [TK]D-Fender | fujin_: You have to imagine what the usres need phones for. Perhaps its a lot of devs who collaborate in office but have no rights to call out, etc. |
| 20:36.46 | lclimber | yes, you are right, my english is not so well, what i am trying to figure out is how many asterisk centrals do i need to allow the 2000 users to comunicate with each other |
| 20:37.01 | [TK]D-Fender | lclimber: Are they in the same building? |
| 20:37.10 | lclimber | yes. in the same building |
| 20:37.44 | [TK]D-Fender | lclimber: Wow... ok, for 100 simultaneous calls, we're not talking a huge deal... I'd go with a single large server with plenty of ram. |
| 20:38.52 | lclimber | hmmm, ok, and does it matter if the users are making calls to analog phones outside the company |
| 20:39.28 | lclimber | or to another asterisk central outside the boulding |
| 20:39.35 | riddlebox | fujin_: I am excited, it will be a rollout for a company, and all their offices, so I will get to maybe make IAX connections between them, or tie them together with T1's |
| 20:39.42 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:41.32 | [TK]D-Fender | lclimber: not that much. Bandwidth outside of your LAN is the real consideration there. |
| 20:41.48 | [TK]D-Fender | lclimber: if we're talking about a limited number of sites. |
| 20:42.06 | km- | I have a cisco phone with disallow=all,allow=g729. It can't make three way conferences, asterisk complains no compatible codecs. |
| 20:42.18 | hmmhesays | I've never had real good luck with iax2 |
| 20:42.38 | fujin_ | km-: have you licensed g729 for asterisk? |
| 20:43.01 | km- | for the purpose of this conversation let us assume yes |
| 20:43.22 | fujin_ | is it working? |
| 20:43.28 | fujin_ | can you disallow all, allow g729 and use voicemail? |
| 20:43.41 | lclimber | but, do i need more than one asterisk box to xomunicate my users???? |
| 20:43.49 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 20:44.10 | Zuchmir | what is the purpose of zapata-channels.conf ? |
| 20:44.25 | km- | yes, I can use voicemail. |
| 20:44.38 | Kobaz | [TK]D-Fender: i monkeied my iax.conf to be as per the example... no go... and i assumed as much |
| 20:44.49 | Zuchmir | <PROTECTED> |
| 20:44.53 | Kobaz | [TK]D-Fender: asterisk isn't handing the call to any dial plans |
| 20:44.53 | *** part/#asterisk lirakis (n=user@69.176.204.146) |
| 20:45.06 | fujin_ | km-: are you globally disallowing all, and allowing g729? |
| 20:45.20 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:45.22 | fujin_ | what codecs does the sip debug show as available? |
| 20:45.39 | *** join/#asterisk d-k-t (n=dt@125.120.138.74) |
| 20:45.54 | [TK]D-Fender | km-: how many licenses? |
| 20:46.26 | fujin_ | G729 issues, providing that you have *actually* licensed it, should be dealt with quite easily by Digium support |
| 20:46.31 | Darthclue | hmmhesays, really? I have better luck with IAX2 than SIP when it comes to bouncing around firewalls. Of course, all of my hard phones are SIP. |
| 20:46.36 | [TK]D-Fender | km->for the purpose of this conversation let us assume yes <-- and perhaps you can elaborate on this.. *cough* |
| 20:46.40 | *** join/#asterisk danuel (i=dan@69.22.167.24) |
| 20:46.58 | danuel | hi |
| 20:47.04 | km- | I'm using intel's implementation of g729 |
| 20:47.06 | Darthclue | ehlo |
| 20:47.29 | danuel | I want to build an asterisk box with the latest hardware..core 2 quad etc.. what new motherboard works well with asterisk? |
| 20:47.51 | fujin_ | intels implementation of g729? |
| 20:48.00 | km- | yes |
| 20:48.02 | fujin_ | That doesn't sound like it'd be supported by Asterisk |
| 20:48.14 | km- | I'm not asking digium support for help here, I'm asking other asterisk users. |
| 20:48.28 | fujin_ | It'd appear that you're doing it wrong. |
| 20:48.39 | fujin_ | The only g729 that works with Asterisk is the Digium one, afaik. |
| 20:48.47 | km- | I appreciate your input. |
| 20:49.08 | km- | The intel g729 implementation works well for everything else but for conferencing |
| 20:49.18 | km- | and I'm wondering if it's a setting I've got wrong somewhere else |
| 20:50.03 | *** part/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 20:50.17 | danuel | anyone? |
| 20:50.28 | km- | Ah. |
| 20:50.30 | km- | It offers toll quality speech at a reasonably low bit rate of 8Kbps. However, it is a rather "costly" codec in terms of CPU processing time, therefore some VoIP phones and adapters (notably the Linksys/Sipura/Cisco brands) can only handle one G.729 call (channel) at a time. This can cause calls to fail if the user attempts to use three-way calling, or place simultaneous calls on both lines of a two-line device, and G.729 is the only allowed codec. |
| 20:50.54 | [TK]D-Fender | km-: could be that Cisco's cont conference in G.729. |
| 20:51.02 | km- | yeah, that's what it's appearing to be. |
| 20:51.07 | [TK]D-Fender | km-: Think I've heard that before. I know its an issue with Linksys |
| 20:51.10 | km- | nuts. |
| 20:51.12 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 20:51.24 | km- | we have this one vp who's remote who demands to use conferencing on his phone |
| 20:51.30 | fujin_ | What's wrong with xLaw? |
| 20:51.34 | Putzz | hey guys is there a way to make audio coming in via sip louder? |
| 20:51.37 | km- | I've already yelled at the bosses, g711 x2 is a lot of bandwidth. |
| 20:51.44 | fujin_ | o_0 |
| 20:51.45 | km- | fujin_: 144kbps (times two calls) |
| 20:52.09 | fujin_ | vp has dialup? |
| 20:52.20 | fujin_ | Putzz: turn up the transmit volume/turn up the handset volume? |
| 20:52.38 | fujin_ | sip/rtp just describes audio, I don't believe you can increase the gain from an asterisk perspective |
| 20:52.39 | km- | vp has cablemodem, but he's not the only person in our enterprise using incoming voip, and we only have bonded T1's here |
| 20:54.29 | Putzz | I have |
| 20:54.39 | [TK]D-Fender | km-: then get him a WRT as a transcoding gateway. |
| 20:54.46 | Putzz | I need to make it louder before it gets to phone somehow |
| 20:54.54 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.37.23) |
| 20:54.55 | [TK]D-Fender | km-: use IAX2 + GSM |
| 21:02.45 | lirakis | I know this is a very broad question, but Im stumped and not sure where to start since i dont know a lot about TDM. Basically, my Sangoma A101 T1 card has failed twice now, which seems to cause my whole Asterisk box to essentially stop working. I have had the switch site operator check that the circuit was good, it was and he said the D channel had gone down. I have NO CLUE what is causing this card to fail. It had been running fine for about 2 mo |
| 21:03.58 | danuel | just rma it with sangoma |
| 21:04.05 | danuel | they have 5 year warranties |
| 21:05.21 | lirakis | danuel: i dont know that it is the card... if i reboot the whole computer it comes back up fine though |
| 21:06.16 | lirakis | if i restart wanpipe, zaptel, asterisk... nothing.. it still doesnt come up... the t1 only seems to come up after a whole system restart |
| 21:07.10 | km- | [TK]D-Fender: oh how I'd love to do that. |
| 21:07.10 | [TK]D-Fender | km-: Or get them a real phone :) |
| 21:07.10 | km- | yeah, we're probably going to force him to get a pots line |
| 21:07.11 | km- | unless he can keep his conference calls to bridges |
| 21:07.23 | km- | I wonder if he knows he can transfer people into his conference bridge |
| 21:09.43 | lirakis | basically... i just dont know how to trouble shoot this. Ive checked /var/log/messages ... /var/log/wanrouter and i see nothing that indicates what caused the failure.. |
| 21:10.07 | jameswf | anyone wanna go halves on a hosted hybrid solution |
| 21:10.14 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 21:10.17 | jameswf | we can be ricjh |
| 21:10.20 | jameswf | *rich |
| 21:12.26 | lirakis | jameswf: probably get richer running gray market routes on gsm gateways ;) |
| 21:14.01 | lirakis | .. ive looked in "the book" and on voip-info .. etc. and im having trouble finding digestable information about digital tdm and asterisk... particularly common trouble shooting (if such a thing exists for digital interfaces on *) |
| 21:14.43 | [TK]D-Fender | lirakis: Call Sangoma support |
| 21:14.48 | blitz-rage | lirakis: that's probably more of a general knowledge thing -- there is a T1 Survival Guide by O'Reilly |
| 21:15.11 | km- | blitz-rage: how's it hangin? haven't talked to you in ages it seems |
| 21:15.13 | blitz-rage | and when you buy hardware from Digium or Sangoma, you should get some support time |
| 21:15.23 | blitz-rage | km-: it has been ages -- I don't even know who you are! :D |
| 21:15.26 | lirakis | blitz-rage: yeah.. ive started looking at it on safari |
| 21:15.52 | km- | blitz-rage: we worked together on the asterisk-doc project like a year and a half ago |
| 21:16.06 | blitz-rage | oh wow... that goes to show just how bad my memory is :) |
| 21:16.12 | km- | haha |
| 21:16.15 | km- | it happens |
| 21:16.45 | km- | I bs'd with bkw a lot, I hear he's on the freeswitch project now? |
| 21:19.30 | *** join/#asterisk snazm (n=snazm@89.243.79.80) |
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| 21:33.15 | jameswf | I am releasing bixtrox 1 today, nothing works but your too stupid to know any better |
| 21:34.04 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 21:34.08 | blitz-rage | you're* |
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| 21:36.36 | *** join/#asterisk KeX (n=KeX@home.fuerstaller.com) |
| 21:36.38 | KeX | hi |
| 21:37.01 | KeX | anyone connected an Asterisk and a Siemens HiPath 4000 via chan_zap using QSIG? |
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| 21:39.02 | lirakis | jameswf: lol |
| 21:39.29 | lirakis | later all |
| 21:39.31 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
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| 21:43.25 | AJaymn | Anyone make a page that would reset an ext voicemail, It would have to reset password to a default one, and remove all messages. |
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| 22:01.39 | hmmhesays | hrm is there some dial flag i have to set for dynamic feature map? |
| 22:02.14 | hmmhesays | er applicationmap |
| 22:04.06 | [TK]D-Fender | hmmhesays, I believe you have to pass it a feature like wW |
| 22:04.25 | [TK]D-Fender | hmmhesays, in addition to the channel var |
| 22:05.13 | hmmhesays | DYNAMIC_FEATURES |
| 22:05.22 | hmmhesays | I have that set, my feature defined in features.conf |
| 22:05.45 | hmmhesays | the wW's are for one touch recording |
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| 22:13.56 | hmmhesays | having a hell of a time with these 501 headsets, it seems they are making the problem |
| 22:19.55 | hmmhesays | [TK]D-Fender: you around I have a question about these m22's |
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| 22:21.15 | [TK]D-Fender | hmmhesays, shoot |
| 22:23.15 | hmmhesays | where should I have the telephone compatibility switch on the m22 when plugged into an ip 501 |
| 22:24.18 | [TK]D-Fender | hmmhesays, not sure.. default was fine IIRC.... |
| 22:24.45 | hmmhesays | i can't freaking figure out this delayed audio problem |
| 22:24.54 | [TK]D-Fender | hmmhesays, the side slide connector was on "5" as well |
| 22:25.24 | hmmhesays | ok |
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| 22:55.00 | ViKing78 | [TK]D-Fender: Do you have any suggestions of what I need to have in my zapata.conf to get 2BCT to work? |
| 22:55.37 | JT | magic=yes |
| 22:56.07 | ViKing78 | JT: Was that for me? magic=yes enables 2BCT? |
| 22:56.39 | JT | yes |
| 22:56.48 | JT | did you check if asterisk actually supports 2BCT? |
| 22:57.17 | ViKing78 | JT: Yes, I'm running the latest libpri/asterisk |
| 22:57.50 | JT | ViKing78: i didn't ask if you were running the latest libpri.. |
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| 22:59.32 | teknoprep | does anyone know of a provider that provides a t.38 line for faxing over voip ... i would use an ATA with this |
| 22:59.53 | ViKing78 | JT: My assumption from the bug entries I've read 2BCT is supported in the latest libpri/asterisk combinations. Am I wrong? |
| 23:00.05 | JT | ViKing78: i think so |
| 23:00.31 | JT | ViKing78: the only work i've heard on 2BCT is with Bristuff, and I'm not aware of exactly what stage it's at |
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| 23:02.24 | ViKing78 | plasterisk01*CLI> show version |
| 23:02.24 | ViKing78 | Asterisk 1.4.13-BRIstuffed-0.4.0-test4 built by pbuilder @ grnetbox on a x86_64 running Linux on 2007-10-10 19:23:45 UTC |
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| 23:03.42 | teknoprep | yay net splits are fun |
| 23:03.42 | JT | ViKing78: well that's bristuff |
| 23:03.42 | ViKing78 | JT: yeah ;) |
| 23:03.42 | JT | ViKing78: worth mentioning. |
| 23:03.42 | JT | and it's a bristuff issue really |
| 23:03.42 | JT | many here will not be familiar with it |
| 23:03.42 | JT | and even less familiar with 2BCT |
| 23:03.42 | teknoprep | i want to be able to outbound from asterisk with t.38... is this possible? |
| 23:03.43 | teknoprep | i am using asterfax for faxing but i want to use t.38... is it implemented yet? |
| 23:03.43 | ViKing78 | JT: No, thanks for the help. Didn't work right off but I'll keep plugging. Any suggestions of who I can talk to to get this to work? I'm willing to contract someone that knows what they're doing. |
| 23:04.44 | ViKing78 | JT: I need this to work with a QSIG switch |
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| 23:04.49 | ViKing78 | I've modded the libpri code to send the eect for a QSIG switch but just can't seem to get it to initiate it. |
| 23:04.50 | JT | ViKing78: kapejod is about the only one who'll know for sure |
| 23:05.01 | JT | teknoprep: no, no |
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| 23:05.39 | ViKing78 | JT: Thanks, I'll pop a pm to him |
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| 23:06.37 | JT | ViKing78: dunno if you'll get a response |
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| 23:09.15 | [TK]D-Fender | ViKing78, http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer |
| 23:09.34 | [TK]D-Fender | "Currently Asterisk(libpri/zaptel) 1.4 fully supports this feature on DMS100 switchtypes only, but there is a patch available for 5ESS and Digium is working on a stable version of that for the 1.4 codebase. As for NI2, Digium has no intentions of supporting 2BCT on it any time soon making this bounty very important." |
| 23:10.22 | JT | what about switchtypes most people actually care about like ETSI? :D |
| 23:10.49 | [TK]D-Fender | JT, Do your telcos allow 2BCT on BRI? |
| 23:10.57 | JT | no idea |
| 23:11.14 | *** join/#asterisk icewaterman (n=immagine@i538740FC.versanet.de) |
| 23:14.26 | [TK]D-Fender | So... who cares :p |
| 23:15.00 | ViKing78 | [TK]D-Fender: I know, but I need it for QSIG. I need it for an IVR app with our phone switch. |
| 23:15.19 | [TK]D-Fender | ViKing78, change your switch :) |
| 23:15.45 | [TK]D-Fender | ViKing78, or make a new bounty, or get coding. Take your pick. |
| 23:16.29 | ViKing78 | [TK]D-Fender: I'm coding on it. Just looking for how to make it work for 5ESS first. Thanks for the help. I'll post a patch if I get it working. |
| 23:16.38 | JT | [TK]D-Fender: BRI is not the only thing here. |
| 23:16.47 | lmadsen | that's a lie |
| 23:17.02 | JT | ? |
| 23:17.05 | lmadsen | :) |
| 23:17.05 | tzafrir_home | ViKing78, BRI as well? |
| 23:17.06 | [TK]D-Fender | lmadsen, don't say that or the 'roos will eat you! |
| 23:17.15 | lmadsen | JT: just joking around |
| 23:17.27 | JT | much more likely to get kicked in the nuts by a roo |
| 23:17.32 | ViKing78 | tzafrir_home: I'm working on PRI for now. |
| 23:17.55 | ViKing78 | tzafrir_home: I don't have a BRI interface to my PBX |
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| 23:24.00 | [TK]D-Fender | BBL |
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| 23:28.37 | teknoprep | damn |
| 23:28.45 | teknoprep | i was going to ask [TK]D-Fender something |
| 23:28.47 | teknoprep | he knows all |
| 23:29.33 | teknoprep | can asterisk work as a t.38 gateway yet? |
| 23:30.19 | anonymouz666 | nops |
| 23:30.24 | nybble | :( |
| 23:30.28 | anonymouz666 | Use callweaver for that |
| 23:31.09 | mog | lies |
| 23:31.17 | mog | you can use spandsp with asterisk |
| 23:31.25 | mog | or you could unless coppice broke it |
| 23:31.55 | JT | err and asterisk supports UDPTL termination? |
| 23:32.02 | JT | and coppice does not break spandsp |
| 23:32.06 | JT | he is the maintainer |
| 23:32.26 | mog | thats my point |
| 23:32.27 | tzafrir_home | mog, spandsp for t.38? |
| 23:32.34 | nybble | thats true, anyone have any opinions on spandsp and rxfax/txfax |
| 23:32.35 | nybble | ? |
| 23:32.45 | mog | if he didnt change to not intentionally work with asterisk then it still works |
| 23:33.03 | mog | personally id use t.38 device |
| 23:33.07 | mog | but thats cause im lazy |
| 23:33.15 | mog | or really avoid faxes all togehter |
| 23:33.17 | tzafrir_home | Speaking of RxFAX and TxFAX: do they actually work? |
| 23:33.19 | mog | er together |
| 23:33.23 | mog | wouldnt know |
| 23:33.26 | JT | it's maintained against callweaver these days |
| 23:33.26 | mog | im told they do |
| 23:33.26 | tzafrir_home | I asked about them a few hours ago |
| 23:33.36 | JT | incompatibilities with asterisk are unintentional |
| 23:33.39 | JT | anyway |
| 23:33.55 | JT | does asterisk's chan_sip support UDPTL termination? |
| 23:34.00 | JT | if not it's a moot point |
| 23:34.34 | tzafrir_home | as in: anybody here actually uses them and can help me figure out why my simple fax from TxFAX to RxFAX does no go through? |
| 23:35.26 | mog | we support udptl in 1.4 |
| 23:35.33 | mog | pass through |
| 23:44.22 | JT | mog: you need termination, not pass through |
| 23:44.36 | JT | mog: otherwise blaming spandsp for not doing T.38 is silly |
| 23:44.40 | mog | i pass through to my t.38 fax machine |
| 23:44.43 | JT | sure |
| 23:44.46 | mog | that was not my goal |
| 23:45.12 | *** join/#asterisk nestAr (i=nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 23:45.40 | nestAr | love freenode |
| 23:46.00 | nestAr | probably have a better time staying connected to EFnet during a DDoS |
| 23:50.48 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
