| 00:00.07 | Miker | fnkmaster, so what to do? |
| 00:00.17 | Jesster | how would I setup zapata.conf for a channel bank that has fxs_ks for ports 1-20 and fxo_ks for 21-24 ? |
| 00:00.25 | jizaymes | Novac, its from Zap |
| 00:00.46 | fnkmaster | Miker: I believe you need a port forwarded range for the RTP stream at work. Because SIP is a kind of stupid protocol. |
| 00:00.58 | Novac | jizaymes: I tink that you could change ths context, in the zap config file, to something not exsisting |
| 00:01.00 | Carp | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3079099650&category=1484 |
| 00:01.05 | Carp | is that a good * server? |
| 00:01.24 | Miker | fnkmaster, that sucks:) |
| 00:01.35 | fnkmaster | Miker: your * box is trying to intiate an RTP stream to the public IP address of your place of work, and the RTP packets are just getting dropped at the router. |
| 00:02.04 | fnkmaster | Miker: at least that's my guess. You can't run a SIP phone behind NAT without changing network configuration parameters because SIP sucks. |
| 00:02.06 | Miker | fnkmaster, i have an asterisk server at the office also but i dont want to register agains this asterisk server |
| 00:02.11 | KillerBee | Carp might be ok for like 25 bucks |
| 00:02.22 | Miker | fnkmaster, can it be use as a proxy? |
| 00:02.37 | Carp | Hmm |
| 00:02.46 | Carp | KillerBee: I have $200, What do you suggest I get? |
| 00:02.59 | marlowe | I have set up an extension i for invalid under my extensions and when I dial an invalid extension it still doesnt play the message? Can anyone provide me with another sample config that works? |
| 00:03.03 | fnkmaster | Miker: yes, in theory, you can have your SIP phone connect to the local * box and have * connect to the remote * machine via IAX2, which circumvents the firewall/NAT issues entirely. |
| 00:03.39 | Miker | fnkmaster, but its a wisip |
| 00:03.44 | Miker | fnkmaster, i use it everywhere |
| 00:03.52 | nitram | fnkmaster: well with the nat stuff enabled in sip.conf and an 7960 that works for me |
| 00:03.56 | *** join/#asterisk telapp (~bonbon@81-86-0-190.dsl.pipex.com) |
| 00:04.28 | nitram | * on public ip, 7960 behind nat, no fancy stuff on nat-firewall |
| 00:04.30 | fnkmaster | nitram: with any port forwards in your network config? |
| 00:04.35 | nitram | yes |
| 00:04.37 | Novac | jizaymes: did you see my suggestion ?? |
| 00:04.43 | Miker | nitram, what kinda router you have? |
| 00:04.57 | nitram | Miker: different dsl-hw routers |
| 00:05.04 | fnkmaster | nitram: right, that's what I said, the key thing is the port forwards for the RTP stream. |
| 00:05.16 | *** join/#asterisk TLC_ (~TLC@h24-68-163-140.du.shawcable.net) |
| 00:05.17 | nitram | fnkmaster: NO port forwarding |
| 00:05.18 | Miker | the problem with a linux router is the uPNP |
| 00:05.27 | Miker | hardware routers use uPNP for that |
| 00:05.44 | nitram | i don't think the cisco 7960 does upnp |
| 00:05.50 | fnkmaster | nitram: hmm, okay, then maybe I'm wrong, maybe you don't need the port forward. |
| 00:06.06 | Miker | nitram, the router |
| 00:06.32 | nitram | Miker: how does upnp work? |
| 00:06.34 | habakuk | anyone know of a linux softphone with dtmf? |
| 00:06.50 | nitram | habakuk: linphone, kphone, sjphone |
| 00:06.53 | Jesster | how would I setup zapata.conf for a channel bank that has fxs_ks for ports 1-20 and fxo_ks for 21-24 ? |
| 00:06.56 | nitram | gnomemeeting |
| 00:07.24 | *** join/#asterisk TestMasTer (~DanJr_Tec@h68-147-106-246.cg.shawcable.net) |
| 00:07.46 | nitram | fnkmaster: tftp nat is not implemented in every hw-router though |
| 00:07.51 | habakuk | nitram just tried kphone, but couldn't get dtmf to work. I found only one obscure reference to it.. any pointers on how to get it to work? |
| 00:07.54 | nitram | so no fancy ringtones |
| 00:08.06 | fnkmaster | nitram: I'm sorry, you're right I don't have any port forwards set up for RTP on my NATed 7960 either |
| 00:08.22 | habakuk | nitram, yeah sip is what I'm looking for |
| 00:08.35 | fnkmaster | I do have a port 5060 port forward, which I thought was necessary to receive calls from the other end, but maybe that's not even necessary |
| 00:08.45 | nitram | habakuk: you need at least v4 |
| 00:09.32 | *** join/#asterisk loko-moko (private@c-67-165-107-230.client.comcast.net) |
| 00:09.36 | fnkmaster | I wonder if I need port 5060 forwarded at all... |
| 00:09.44 | nitram | what gave/gives me a headache is firewall without nat |
| 00:09.46 | habakuk | nitram ah ok |
| 00:09.58 | jizaymes | hrm |
| 00:10.07 | jizaymes | i set the context to something different and it still hangs up |
| 00:10.09 | jizaymes | er |
| 00:10.10 | jizaymes | picks up |
| 00:10.17 | nitram | where i have a public ip but a syn-filter |
| 00:10.37 | nitram | so i can make outgoing connections |
| 00:11.04 | nitram | so my 7960 logged in fine, i could make calls, but not be called |
| 00:11.28 | JerJer | NAT? |
| 00:11.35 | nitram | no |
| 00:11.38 | fnkmaster | anybody have any experience with the Cisco 12SP+ phones? I can't even figure out how to tell the damned thing where to find a TFTP server (my DHCP server doesn't supply TFTP info) |
| 00:11.40 | jizaymes | nat=yes in sip.conf |
| 00:11.42 | JerJer | firewall? |
| 00:12.00 | JerJer | fnkmaster: i think u gots to have the right firmware on 'em |
| 00:12.16 | nitram | JerJer: yes, firewall but public ip on both ends |
| 00:12.18 | Novac | jizaymes: didnīt you want * to ignore incomming calls from zap |
| 00:12.21 | nitram | (university) |
| 00:12.40 | fnkmaster | JerJer: I know, but I have to at least get it to talk to my TFTP server to put firmware on em (I assume they're like the 7960 in that sense) |
| 00:12.40 | JerJer | nitram: are you allowing udp port 5060? |
| 00:12.50 | Novac | jizaymes: then is should hang up |
| 00:12.53 | JerJer | make a DNS entry |
| 00:13.00 | jizaymes | yes |
| 00:13.01 | nitram | JerJer: i have no access to university's fw :/ |
| 00:13.03 | jizaymes | i have two Zap interfaces |
| 00:13.09 | jizaymes | i want the second for outgoing only |
| 00:13.23 | JerJer | SelsiusCM1 |
| 00:13.31 | JerJer | come on.....run your own |
| 00:13.38 | bkw_ | haha |
| 00:13.39 | JerJer | make a private network |
| 00:13.48 | fnkmaster | hehehe this sounds like fun |
| 00:13.50 | jizaymes | in zapata.conf i have this |
| 00:13.50 | jizaymes | context=default |
| 00:13.50 | jizaymes | channel => 1 |
| 00:13.50 | jizaymes | context=lame |
| 00:13.50 | jizaymes | channel => 2 |
| 00:14.09 | jizaymes | i've tried with lame existing but being empty, aswell as lame not existing |
| 00:14.12 | jizaymes | it still answers |
| 00:14.12 | fnkmaster | gotta love it, a 15 dollar phone that requires about 500 dollars worth of time to get working |
| 00:14.25 | JerJer | jizaymes: do you have immediate=yes in there? |
| 00:14.32 | nitram | JerJer: i know i could tunnel in there via my laptop and then connect... but i'd like it to work directly |
| 00:14.42 | JerJer | bleh |
| 00:14.47 | jizaymes | no |
| 00:14.49 | jizaymes | it says immediate=no |
| 00:14.55 | jizaymes | immediate=no |
| 00:15.46 | telapp | can i use #includes within iax.conf? |
| 00:15.53 | Novac | jizaymes: maybe if you define a context with a hangup command |
| 00:16.16 | jizaymes | true |
| 00:16.17 | jizaymes | but thats lame |
| 00:16.27 | JerJer | ? |
| 00:16.36 | JerJer | jizaymes: esplain whats going on |
| 00:16.38 | JerJer | again |
| 00:16.59 | loko-moko | How does call forwarding on the Cisco 7960 work? I hit the call forwarding button, type in my number, but when I call that extension it just sends me to voicemail saying im un available |
| 00:17.05 | jizaymes | ok |
| 00:17.08 | jizaymes | have 2 zap interfaces |
| 00:17.12 | jizaymes | Zap/1 == incoming only |
| 00:17.16 | jizaymes | Zap/2 == outgoing only |
| 00:17.21 | jizaymes | however |
| 00:17.29 | jizaymes | calls to Zap/2 answer and bring to the menu |
| 00:17.30 | marlowe | For some reason when I try to transfer a call from my phone to another phone, and if I enter an invalid extension the call just ends up getting lost. I tried making a routine to play an invalid gsm file, which it does - But it plays it to the caller and not to me... Is there any other way I can prevent from accidently transferring to an invalid extension? |
| 00:17.55 | unixdawg | chan_iax2.c: In function `try_firmware': |
| 00:17.55 | unixdawg | chan_iax2.c:883: error: storage size of `stbuf' isn't known |
| 00:17.55 | unixdawg | chan_iax2.c:890: warning: implicit declaration of function `stat' |
| 00:17.55 | unixdawg | chan_iax2.c:896: warning: implicit declaration of function `S_ISDIR' |
| 00:17.55 | unixdawg | chan_iax2.c:883: warning: unused variable `stbuf' |
| 00:17.56 | unixdawg | gmake[1]: *** [chan_iax2.o] Error 1 |
| 00:17.58 | unixdawg | gmake[1]: Leaving directory `/usr/home/asterisk/asterisk/channels' |
| 00:18.00 | unixdawg | gmake: *** [subdirs] Error 1 |
| 00:18.04 | marlowe | Possible, in this function to call back the person that tried transferring the call... So they can attempt to transfer again? |
| 00:18.16 | jizaymes | ok Novac |
| 00:18.18 | jizaymes | that worked |
| 00:18.20 | jizaymes | but its still lame |
| 00:18.22 | jizaymes | cause it makes a click |
| 00:18.23 | JerJer | jizaymes: then something is telling it to answer |
| 00:18.44 | jizaymes | ok what would do that |
| 00:18.45 | jizaymes | heh |
| 00:18.51 | Damin | marlowe: can you use supervised transfewr? |
| 00:19.09 | marlowe | Damin: Don't think that works with these phones.. The GS phones? |
| 00:19.09 | JerJer | something like exten => s,1,Answer |
| 00:19.19 | jizaymes | in the [default] context i do |
| 00:19.29 | jizaymes | but that channel is bound to the [lame] context |
| 00:19.46 | Damin | marlowe: Oh yeah.. the only way to get that to work with the GrandStream is to smash it with a hammer and get a SNOM or Cisco unit! ;) |
| 00:19.49 | unixdawg | any innput on my error |
| 00:19.57 | marlowe | Damin: Oh ok. :-/ |
| 00:20.29 | jizaymes | also, who has the ability to port vonage phone #'s and give me iax for a fee |
| 00:20.34 | Damin | marlowe: I wish the GrandStream allowed you to configure the type of transfer that you could use. |
| 00:20.35 | *** join/#asterisk JazzInc (~Unknown@61.29.44.74) |
| 00:20.51 | Damin | marlowe: It's just a software configuration issue really.. |
| 00:20.57 | JerJer | jizaymes: depends where they are at |
| 00:21.00 | scud | hey whats the link to the iax protocol.pdf? |
| 00:21.05 | jizaymes | its in zapata.conf |
| 00:21.12 | *** join/#asterisk amiramir (amiramir@adsl-216-220-113-6.bway.net) |
| 00:21.23 | Damin | marlowe: Unfortunately, you don't have control of the software. ;) |
| 00:21.26 | jizaymes | context=default |
| 00:21.27 | jizaymes | channel => 1 |
| 00:21.27 | jizaymes | context=lame |
| 00:21.27 | jizaymes | channel => 2 |
| 00:21.32 | marlowe | So is there any way to capture within asterisk the extension that attempted to transfer the cal.. So maybe asterisk can then just issue a dial statement and call that extension back? |
| 00:21.38 | marlowe | so the call doesnt get lost? |
| 00:21.57 | Damin | marlowe: Perhaps you would be better off using Call Park? |
| 00:22.13 | Novac | jizaymes: what about exten => s,1,NoOp |
| 00:22.19 | marlowe | Never been a fan of call park, I'll just live with it... |
| 00:22.32 | marlowe | I can easily transfer the caller into the main ivr at least. |
| 00:22.38 | marlowe | So I dont hang up on them |
| 00:22.45 | jizaymes | ok |
| 00:22.48 | jizaymes | let em try that |
| 00:22.49 | jizaymes | me |
| 00:22.52 | Damin | exten => s,1,Hangup |
| 00:22.55 | marlowe | Theres gotta be a way to issue a call back though |
| 00:23.02 | marlowe | I dont want to hangup on them :) |
| 00:23.24 | jizaymes | works like a champ Novac |
| 00:23.25 | jizaymes | thanks |
| 00:23.41 | Novac | jizaymes; You are welcome |
| 00:24.03 | unixdawg | ok trying make again |
| 00:24.40 | Novac | jizaymes: just for the fun of it, were are you from ?? |
| 00:24.48 | jizaymes | new jersey |
| 00:24.54 | marlowe | me too |
| 00:25.00 | jizaymes | word up neighbor |
| 00:25.11 | marlowe | what up :) |
| 00:25.14 | marlowe | where in nj |
| 00:25.42 | jizaymes | rockaway |
| 00:25.42 | Novac | jizaymes: ever heard about denmark ?? |
| 00:25.55 | jizaymes | yeah |
| 00:26.05 | marlowe | never heard of it |
| 00:26.07 | marlowe | lol |
| 00:26.17 | Novac | that were I am from |
| 00:26.36 | jizaymes | hrm |
| 00:26.41 | marlowe | lol |
| 00:26.41 | jizaymes | ok next problem |
| 00:26.44 | jizaymes | when I dial an extension |
| 00:26.46 | jizaymes | and hangup |
| 00:26.52 | jizaymes | * doesnt recognize the hangup |
| 00:26.58 | jizaymes | and continues to ring |
| 00:27.04 | marlowe | I just had that prob with my grandstream phone |
| 00:27.13 | jizaymes | i'm using Dial(SIP/101,15,tTm) |
| 00:27.31 | Novac | jizaymes: from another SIP |
| 00:27.57 | jizaymes | nope |
| 00:27.58 | jizaymes | from Zap |
| 00:27.59 | jizaymes | :) |
| 00:28.03 | jizaymes | i have |
| 00:28.16 | jizaymes | exten => h,1,Hangup in every context |
| 00:28.42 | *** join/#asterisk FuzzyCat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 00:28.43 | *** join/#asterisk Flatcat (~ScaredyCa@c44095.upc-c.chello.nl) |
| 00:29.14 | Novac | jizaymes: what kind of hardware do Y have |
| 00:29.22 | jizaymes | Zap wcfxo |
| 00:29.29 | jizaymes | all analog |
| 00:29.33 | jizaymes | using fxsks |
| 00:29.40 | Novac | jizaymes ok |
| 00:30.08 | Jesster | how would I setup zapata.conf for a channel bank that has fxs_ks for ports 1-20 and fxo_ks for 21-24 ? |
| 00:30.34 | jizaymes | signalling=blah |
| 00:30.37 | jizaymes | channel => 1 |
| 00:30.40 | jizaymes | ... -20 |
| 00:30.43 | jizaymes | signaling=blah |
| 00:30.46 | jizaymes | channel => 21 |
| 00:30.46 | jizaymes | ... |
| 00:30.47 | jizaymes | 24 |
| 00:30.50 | jizaymes | i would think |
| 00:30.54 | jizaymes | never done it though |
| 00:31.00 | Jesster | well it's worth a shot! thanks. |
| 00:31.04 | jizaymes | actually |
| 00:31.11 | jizaymes | it might be better off in /etc/zaptel.conf |
| 00:31.24 | jizaymes | fxsks=1-20 |
| 00:31.30 | jizaymes | fkoks=21-24 |
| 00:31.41 | Jesster | yea I tried that, and it complained, lemme get the error. |
| 00:31.52 | file | errors are good. |
| 00:31.56 | jizaymes | actually |
| 00:31.58 | file | for fixing stuff that is |
| 00:32.04 | jizaymes | shouldnt you be multiplexing that into a t1 or something |
| 00:32.56 | Jesster | i think it will eventually, Im doing the setup before the T1 is setup, so right now were using a channel bank |
| 00:35.26 | Novac | anyone used the wireless Zyxel phone ?? |
| 00:35.27 | Jesster | hrm, i guess those errors were a figment of my imagination, coulda swore zaptel.conf had that before i messed it up with zapata.conf |
| 00:35.57 | Jesster | now i just get unable to create channel of type 'Zap'.. but im thinkin that's a seperate issue. |
| 00:36.02 | nitram | yay :) |
| 00:36.51 | nitram | quality was perfect :) |
| 00:37.01 | nitram | THANK YOU ASTERISK :) |
| 00:38.59 | jizaymes | ok |
| 00:39.00 | jizaymes | question |
| 00:39.01 | jizaymes | 'g' -- goes on in context if the destination channel hangs up |
| 00:39.05 | jizaymes | thats from Dial() |
| 00:39.14 | jizaymes | does that mean it'll proceed to s,2 |
| 00:39.16 | jizaymes | or s,103 |
| 00:39.36 | jizaymes | q |
| 00:39.42 | unixdawg | ok the gs has a 4mb disk on a chip flash drive |
| 00:42.19 | KillerBee | damn voicepulse down again? |
| 00:43.26 | unixdawg | no |
| 00:43.32 | file | unixdawg: a doc? ooh |
| 00:44.21 | unixdawg | the mx chip is a doc |
| 00:44.27 | unixdawg | vp is up |
| 00:44.53 | file | you should take some pics and put 'em up |
| 00:45.32 | *** join/#asterisk bobman (~bobman@68-169-206-81.agstme.adelphia.net) |
| 00:45.50 | HoopyCat | unixdawg: we got the budgetone working with a plantronics amp, btw... slightly adjusted pinout on the cable to the amp... |
| 00:46.31 | unixdawg | ? |
| 00:46.46 | unixdawg | ok |
| 00:47.20 | HoopyCat | unixdawg: i'll keep you informed as i hear back from plantronics. :-) |
| 00:47.31 | unixdawg | ok |
| 00:48.25 | gl[zZz] | good n8 to you all ... |
| 00:50.11 | *** join/#asterisk fnkmaster (fnkmaster@66-108-224-80.nyc.rr.com) |
| 00:51.18 | JerJer | man its early |