| 00:03.41 | *** join/#asterisk MarkD__ (~mark@190.197.85.16) |
| 00:05.06 | MarkD__ | hi everyone... I'm too newb to figure this out myself.. I have a few 7920 phones registered via SCCP (chan_sccp), but every number they call is sent as if they're calling operator |
| 00:05.18 | MarkD__ | could someone give me a hint why? |
| 00:23.58 | WIMPy | MarkD__: I guess you need a WaitExten at s in the phones context. Or maybe not having s might also help. |
| 00:24.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yndlzhuarmvrguyx) |
| 00:37.58 | MarkD__ | in the sccp config file I'd add that?? |
| 00:38.55 | WIMPy | No, that's your extensions.conf. |
| 00:39.35 | MarkD__ | and if I'm using freepbx?? |
| 00:39.46 | WIMPy | ~freepbx |
| 00:39.46 | infobot | [~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 |
| 00:39.51 | MarkD__ | I have the SCCP/# and it work to recieve |
| 00:40.10 | MarkD__ | I did, and there has not been a single response |
| 00:40.39 | WIMPy | Then you will need more patience. |
| 00:41.20 | MarkD__ | ok, yes, I know.. I'm at a client's site wanted to get it done and leave, but might have to come back |
| 00:42.41 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 00:42.45 | elliot98 | slips in |
| 00:43.24 | elliot98 | upgraded to 1.8 and getting dahdi/i1/5551212 on incoming PRIs |
| 00:43.35 | elliot98 | is the i1 something new? |
| 00:43.48 | elliot98 | usually, the channel group should be there |
| 00:43.51 | WIMPy | span 1 |
| 00:45.38 | elliot98 | so it doesn't show the channel group as in 1.4? |
| 00:46.22 | WIMPy | I can't really remember how it was back then. It used the group? |
| 00:46.39 | WIMPy | But channels can be in multiple groups. |
| 00:48.04 | elliot98 | it used to list the DAHDI channel, like, "DAHDI/19-1" |
| 00:48.18 | elliot98 | now it lists the span |
| 00:48.56 | WIMPy | That makes more sense. |
| 00:49.12 | elliot98 | seems the callerid is there instead |
| 00:50.20 | *** join/#asterisk oneadvent (~oneadvent@ip68-105-63-31.pn.at.cox.net) |
| 00:50.43 | oneadvent | hi, i would like to get the number the fax was sent to as a variable in FaxDispatch, anyone know how to do that |
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| 01:54.12 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 01:54.34 | omani | when I set a default soruce IP of all OUTGOING packages from asterisk to WAN. do I have to use externip in sip.conf then? |
| 01:55.27 | omani | like I have asterisk node0 and asterisk node1 (both high availablity). those two share one virtual IP address. like 192.168.1.1. my firewall shall NAT everything from 192.168.1.1 outgoing. |
| 01:55.37 | omani | so I say externip=192.168.1.1 in sip.conf right? |
| 01:56.32 | WIMPy | You have to set externip if the IP Asterisk is reachable by, isn't the one used on the outgoing interface. I.e. when you're behind NAT. |
| 01:57.01 | omani | I am behind nat |
| 01:57.09 | omani | but dont wanna have two IP. |
| 01:57.29 | omani | on failover, asterisk shall be reachable by firewall (which is doing all the natting for me) to 192.168.1.1 |
| 01:57.48 | omani | instead of 192.168.1.2 when node0 is master and 192.168.2.3 when node1 is master |
| 01:58.39 | omani | I have to use one IP also, because my firewall settings allow me to set only one IP in DMZ. |
| 01:58.43 | WIMPy | So you have two default gateways? |
| 01:58.48 | omani | no only one |
| 01:58.56 | omani | ah sry |
| 01:59.04 | omani | 192.168.2.3 is 192.168.1.3 actually |
| 01:59.51 | WIMPy | I don't get the full picture, yet. |
| 01:59.56 | omani | firewall dmz set ip is: 192.168.1.1. node0 has x.2 node1 has x.3 but both share virtual IP (with carp protocoll) 192.168.1.1 |
| 02:00.29 | omani | so if node0 fails. node1 is becoming master and shares the virtual ip of 192.168.1.1 |
| 02:00.35 | omani | now got it? |
| 02:00.44 | WIMPy | Those nodes are the ones running Asterisk? |
| 02:00.47 | omani | yes |
| 02:00.53 | WIMPy | ok |
| 02:01.16 | omani | same asterisk. same configuration. just two different REAL ip. but share the same virtual ip. |
| 02:01.20 | WIMPy | But yes, yu need to set externip to the public IP. |
| 02:01.28 | omani | and what is the public IP then? |
| 02:01.36 | omani | Im still natting in DMZ |
| 02:01.48 | WIMPy | Whatever your public IP is. |
| 02:01.49 | omani | have no real public WAN IP reserved for it |
| 02:02.17 | WIMPy | So how do you expect to reach those Asterisk nodes? |
| 02:02.51 | omani | when asterisk is registering through firewall (to sip provider). firewall opens the port 5060. |
| 02:03.05 | omani | so both bidirectional communication is possible |
| 02:03.45 | WIMPy | So you don't expect/want Asterisk to be reachable from the internet? |
| 02:03.50 | omani | no |
| 02:03.58 | WIMPy | ok |
| 02:03.59 | omani | who wants that? |
| 02:04.21 | WIMPy | What's the point of using VOIP otherwise? |
| 02:04.30 | omani | now the only question is: how do set the source ip of both asterisk server to 192.168.1.1 |
| 02:04.46 | WIMPy | But you still need the external IP for the RTP stream to reach you, unless your provider takes care of that. |
| 02:04.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 02:05.10 | WIMPy | The internal IP doesn't matter. |
| 02:05.17 | omani | if I register to sip provider he already gets my IP |
| 02:05.28 | omani | look |
| 02:05.46 | omani | asterisk 192.168.1.1 -> firewall NAT TO EXTERNAL IP -> sip provider |
| 02:05.55 | WIMPy | Yes, that's good enough for SIP, but RTP is negotiated inband, just like FTP or DCC. |
| 02:06.07 | omani | sip provider -> firewall NAT BACK TO INTERNAL IP/PORT -> asterisk at 192.168.1.1 |
| 02:06.31 | omani | and? I'm in dmz you forget that? |
| 02:06.44 | WIMPy | That's the issue. |
| 02:06.45 | omani | all ports are open and reachable |
| 02:06.55 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:07.27 | WIMPy | So it's not a DMZ but a rather a forward all ports thing? |
| 02:07.51 | omani | well, right now its a port forwarding / port triggering mix thing |
| 02:07.57 | omani | but I have low quality |
| 02:08.04 | omani | stuttering in speech |
| 02:08.07 | omani | RTP |
| 02:08.18 | omani | and the only way it works is to set asterisk in DMZ |
| 02:08.37 | omani | but my DMZ just takes one IP. its a firewall router (so dmz is like exposed HOST thing) |
| 02:08.40 | omani | you know |
| 02:08.43 | omani | DMZ lite :) |
| 02:08.58 | WIMPy | That's not DMZ at all. |
| 02:09.06 | omani | and I thought, since I can only use one IP there and I want to do failover. share a virtual IP. |
| 02:09.12 | omani | I know |
| 02:09.16 | omani | but I can't help |
| 02:10.47 | omani | so the question short is: how do I force asterisk to use either eth1:ucarp rather than eth1 or force it to have a source IP of 192.168.1.1 |
| 02:10.58 | omani | which is in both cases the virtual IP |
| 02:11.24 | WIMPy | I'm not sure if bindaddr will set it. |
| 02:11.48 | omani | but can't use bindaddr. since I have two subnets |
| 02:11.51 | WIMPy | Usually Asterisk doesn't set the IP but lets the routing table do it. |
| 02:11.56 | omani | and all phones are in 192.168.100.X |
| 02:12.48 | WIMPy | Pretty fishy. |
| 02:12.52 | omani | :/ |
| 02:13.03 | WIMPy | You have to make that virtual IP the only default route. |
| 02:13.34 | omani | its a virtual IP both machines have internal |
| 02:13.40 | omani | I can't route |
| 02:14.03 | WIMPy | Game over. |
| 02:14.08 | omani | :'( |
| 03:24.07 | bdfoster | suggestion: FreeSWITCH |
| 03:24.18 | bdfoster | multiple sofia profiles |
| 03:24.24 | bdfoster | problem solved |
| 03:24.27 | bdfoster | NEXT! |
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| 04:55.23 | gajini | Hi, How to install digium PRI cards with mobicent jSS7? Please guide me. |
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| 06:14.38 | Jo_Bell | Hi. I have to implement a SIP client in software to connect to Asterisk server. Any document containing the sequences of protocol handshake needed between client and Asterisk? Any guidelines are also welcome. This is for generic reading purpose |
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| 06:19.31 | ectospasm | Jo_Bell: RFC 3261 |
| 06:20.12 | ectospasm | RFC 3665 has some basic call flow examples |
| 06:25.36 | Jo_Bell | ectospasm: thanks. Let me check |
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| 06:58.26 | fling | hello! |
| 06:58.40 | Jo_Bell | Suppose, I have a SIP client registered with Asterisk Server. I want to make calls to another SIP client registered with a different SIP server. Can such calls between clients connected to different servers possible in Asterisk? |
| 06:58.44 | fling | can you please help me to setup? |
| 06:59.07 | slav3_kitten | attention british people. i wish to wish you all a happy "You sucked at war" day! |
| 06:59.51 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 06:59.54 | vader-- | hello |
| 06:59.56 | vader-- | anyone awake? |
| 07:06.23 | vader-- | kinda in a bind |
| 07:06.50 | vader-- | trying to get an astribank 8 fxs/8fxo setup in asterisk 1.8 and my dahdi channels are showing up only as a psuedo |
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| 07:09.59 | vader-- | hi bulk |
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| 07:25.32 | gajini | #mobicents |
| 07:33.12 | bulkorok | hi |
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| 07:38.23 | awk | Hi, please ca someone check this pastebin for me, and see what could be wrong with my rpmbuild http://pastebin.com/TYpcN9Nm |
| 07:38.41 | awk | I'm so close, just not sure why it's failing ? |
| 07:39.25 | Samuel_CTX | awk: did you check this?: /var/tmp/rpm-tmp.8561: line 181: --with-curl=/usr: No such file or directory |
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| 07:41.48 | awk | Samuel_CTX curl is installed, let me hash that line out and see if it compiles, thanks... |
| 07:41.58 | awk | or maybe curl isn't part of /usr |
| 07:42.18 | awk | it's /usr/bin/ |
| 07:42.29 | awk | let me create a sym link and see if that fixes it |
| 07:43.31 | awk | have to get this amr codec working, the transcoding on the audiocodes isn't working correctly. so 1 step at ta time |
| 07:43.32 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 07:43.50 | elliot98 | where can I find the official documentation about setting up encrypted SIP and RTP |
| 07:43.51 | elliot98 | ? |
| 07:45.47 | vader-- | hmmm anyone a semi expert at diagnosing dahdi problems? |
| 07:45.53 | vader-- | i can't get dahdi to create any channels |
| 07:45.57 | vader-- | all i get psuedo |
| 07:46.32 | elliot98 | what does "pri show spans" show? |
| 07:47.32 | elliot98 | is there any minimum version of DAHDI for 1.8? |
| 07:49.12 | vader-- | no such command |
| 07:50.05 | Samuel_CTX | vader--: what does lsusb and lsdahdi tell? |
| 07:50.42 | vader-- | it shows the board |
| 07:50.48 | vader-- | and the spans |
| 07:51.22 | vader-- | is there a way to rerun the asterisk install |
| 07:51.28 | vader-- | and make sure dahdi gets reinstalled properly? |
| 07:52.05 | Samuel_CTX | you install dahdi seperate from asterisk (at least its how I do it) |
| 07:52.23 | vader-- | i installed dahdi after i installed asterisk |
| 07:52.31 | vader-- | so i reran the setup |
| 07:52.38 | vader-- | im thinking that might of bunked something |
| 07:52.41 | Samuel_CTX | you can stop asterisk, then restart the dahdi service (in your prefered way) and then restart asterisk, and asterisk should see it |
| 07:54.58 | *** join/#asterisk emate (~marcin@81.219.183.142) |
| 07:55.12 | emate | Hi guys |
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| 08:07.00 | vader-- | hmmm |
| 08:07.15 | vader-- | i tried ./configure again then menuconfig and it won't open menuconfig |
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| 08:12.50 | emate | I'm wondering is there any way to get the destination number of incoming call? I have 100 numbers and want to know which number have the most incoming calls. How can i get this stats? |
| 08:14.40 | Samuel_CTX | emate: 1) configure CDR: https://www.asterisk.org/doxygen/trunk/Config_cdr.html ( http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql ) - 2) query your CDR |
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| 08:16.40 | fling | asterist is not listening :5060 when started with initscript. How to fix it? |
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| 08:17.33 | fling | fulcan: hey |
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| 08:26.42 | Alphacube | Is there a way to get Asterisk to read CallerId for a user from LDAP? |
| 08:28.12 | Alphacube | I have seen references to a application called LDAPGet but thats a third party application. Is there another way? |
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| 08:29.49 | din3sh | hello all |
| 08:30.50 | vader-- | hmm it's crazy |
| 08:32.03 | vader-- | http://pastebin.ca/2166969 |
| 08:37.29 | emate | Samuel_CTX: I have cdr configured in Trixbox, but i haven't found inbound number in any of columns in mysql. |
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| 08:41.27 | fling | my sip module is not loaded if I start asterisk with iniscript |
| 08:41.37 | fling | Help me please to fix it |
| 08:59.13 | elliot98 | if upgrading asterisk, do you need to reinstall DAHDI? |
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| 09:04.09 | din3sh | elliot98: not really , if you already have latest version of dahdi installed |
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| 10:33.32 | elliot98 | is there is a way to save SIP debugs from the asterisk console into a pcap or external file? |
| 10:47.28 | fling | elliot98: I want it too |
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| 11:07.57 | elliot98 | what is the issue with "hairpin" calls in promisredir? |
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| 11:22.31 | elliot98 | a bit confused about sip.conf t1 timers, it seems a different than the rfx |
| 11:22.33 | elliot98 | *rfc |
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| 11:23.44 | elliot98 | according to the RFC 3261, the final response is 64*t1, whereas sip.conf states that the max time for a provisional response? |
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| 11:29.17 | TSM2 | there is a guide on how to capture PRI/T1 etc but using wanpipe for sangoma |
| 11:29.23 | TSM2 | lucky i have sangoma cards |
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| 11:36.59 | _omer | I have a question: if my device and Asterisk Support TLS Protocol. does it mean that device can connect to Asterisk if SIP is blocked by Internet service provider? |
| 11:37.14 | Chainsaw | _omer: It depends on how the block is implemented. |
| 11:37.27 | Chainsaw | _omer: But I suppose it is worth a go. |
| 11:39.37 | _omer | Chainsaw: Lets suppose SIP is blocked. |
| 11:39.59 | _omer | SIP Protocol...Not sip port. |
| 11:40.48 | _omer | Internet service provider has blocked SIP Protocol, No matters what the port is. |
| 11:42.52 | Chainsaw | _omer: Then yes, all that should be visible is the STARTTLS. |
| 11:43.10 | Chainsaw | _omer: Doesn't mean they won't block that of course. |
| 11:45.31 | _omer | Chainsaw: so TLS is not a solution to make voip working where SIP is blocked...right ? |
| 11:46.22 | _omer | I can understand that TLS is for hand shaking. |
| 11:47.36 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 11:53.55 | *** join/#asterisk wonderworld (~ww@dsdf-4db53dcb.pool.mediaWays.net) |
| 11:55.46 | *** join/#asterisk dym (~patrick@unaffiliated/dym) |
| 11:56.55 | dym | Hey - im having trouble with attended transfer. I have set the atxfer in features.conf to *2. I have a number that is redirected to a cellphone - so when i call that number, accept on the cell and press *2 nothing happens. When Dialing with tT and pressing # i get a dialing prompt, but only for blind transfer. any idea how to figure this? |
| 11:58.44 | elliot98 | does hinting work with SIP realtime friends? |
| 12:00.39 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 12:06.13 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 12:17.48 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:23.51 | fling | how can I debug why my sip module is not autoloading? |
| 12:25.03 | WIMPy | Yur sip.conf may be bad. |
| 12:25.12 | [TK]D-Fender | Shouldn't be |
| 12:25.18 | [TK]D-Fender | otherwise ti wouldn't load manually either |
| 12:25.30 | [TK]D-Fender | modules.conf <- |
| 12:26.24 | WIMPy | If it works manually, no. |
| 12:27.25 | fling | it works manually |
| 12:27.37 | *** join/#asterisk philfine (~tino@a95-92-251-229.cpe.netcabo.pt) |
| 12:27.44 | philfine | Hi everyone |
| 12:27.47 | fling | also autoload works when I start asterisk with asterisk -cvvv |
| 12:28.09 | fling | but autoload does not work when I star asterisk with initscript |
| 12:28.34 | WIMPy | Does the script change user? |
| 12:28.35 | fling | but I still can connect to it and load sip manually :p |
| 12:28.48 | WIMPy | Interesting. |
| 12:29.08 | fling | /etc/init.d/asterisk > http://dpaste.com/767270/ |
| 12:29.18 | WIMPy | philfine: You cam to hear about AGI? ;-) |
| 12:29.18 | philfine | How can I permit a list of incoming caller ids to fall in a different context, I mean the calls comes from a single channel buit depending on the caller id allow different types of service |
| 12:30.02 | WIMPy | philfine: There is the exten => exten/cid thing. |
| 12:30.18 | [TK]D-Fender | phiNo, calls come in against a given peer into the specified context. If you want to split by CID, you have to do it in dialplan logic starting from that point |
| 12:30.59 | [TK]D-Fender | philfine, ^ |
| 12:31.24 | philfine | So you will have to have as many lines in the dial plan as incoming call ids right ? |
| 12:31.36 | philfine | possible incoming call ids |
| 12:31.45 | WIMPy | Or use something external. |
| 12:32.17 | philfine | There is no actual way to match against a list of elements |
| 12:32.23 | oneadvent | hi, i would like to get the number the fax was sent to as a variable in FaxDispatch, anyone know how to do that |
| 12:32.34 | WIMPy | Call a script. |
| 12:32.55 | [TK]D-Fender | philfine, Depends |
| 12:32.56 | WIMPy | Or use the number as the key to a database. |
| 12:33.34 | philfine | Ok |
| 12:33.37 | [TK]D-Fender | philfine, If EVERYONE is special and you expect a ton of different people, well you need a list somewhere |
| 12:34.14 | fling | WIMPy: [TK]D-Fender: is it a bug? or just a bad initscript? also nothing in /var/log/asterisk/messages if I start with initscript instead of asterisk -c |
| 12:34.19 | philfine | But there is no way to define those list in extensions.cong |
| 12:34.21 | philfine | conf |
| 12:34.36 | philfine | Unless you create rules |
| 12:34.49 | [TK]D-Fender | fling, "ls -la /etc/asterisk" <- PB it |
| 12:34.57 | WIMPy | philfine: That's the way. Or use a database lookup. |
| 12:34.58 | fling | [TK]D-Fender: WIMPy: sip.conf and extensions.conf from here > http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
| 12:35.11 | [TK]D-Fender | philfine, Well, there aer ways... |
| 12:35.26 | fling | WIMPy: [TK]D-Fender: http://dpaste.com/767194/ |
| 12:35.35 | WIMPy | fling: You need a verbose/debug log of Asterisk starting. |
| 12:35.50 | fling | WIMPy: how to enable it? |
| 12:36.10 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 12:36.12 | fling | ^ other files are samples |
| 12:36.19 | WIMPy | fling: You can turn them up in asterisk.conf and capture the output by having asterisk -R running befor starting it via init script. |
| 12:36.32 | fling | WIMPy: great! |
| 12:36.34 | Dennisvj | philfine, I do something similar. We have ten numbers that need to be routed no matter if we are closed or not. That is done on extensions.conf logic by matching caller-id. If no match has been found, the dialplan goes along, otherwise it jumps to the instruction for the special numers (dial specific extension) |
| 12:37.10 | fling | WIMPy: what option do I need? |
| 12:37.12 | elliot98 | faxing is not switching to T38...does there need to be special setting in sip.conf (besides the uptl38 settings)? |
| 12:37.49 | [TK]D-Fender | fling, I'm wondering if you're trying to run * as NON-ROOT as your configs are owned by root |
| 12:38.01 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:38.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:38.08 | [TK]D-Fender | fling, since you can run it MANUAL and it works, that seems suspect |
| 12:38.31 | fling | uid=101(asterisk) gid=120(asterisk) группы=120(asterisk),20(dialout) |
| 12:38.38 | [TK]D-Fender | fling, that's the problem |
| 12:38.40 | fling | [TK]D-Fender: I have asterisk user! |
| 12:38.44 | [TK]D-Fender | etc/asterisk is all ROOT |
| 12:38.51 | [TK]D-Fender | your cofigs have the wrong OWNER |
| 12:39.31 | fling | oh! really? why config files should have different uid? |
| 12:39.34 | WIMPy | Well it's good if not everyone can read them, but Asterisk itself should be ablte to. |
| 12:39.44 | fling | I will now try to start as asterisk |
| 12:39.57 | [TK]D-Fender | fling, Hey, they're YOUR files... you tell us. In the meantime, jsut chmod the folder & files |
| 12:40.04 | [TK]D-Fender | ... CHOWN rather |
| 12:40.07 | fling | lol |
| 12:40.24 | fling | -rw-r----- is the problem :p |
| 12:40.26 | fling | thanks |
| 12:40.37 | [TK]D-Fender | No, the OWNER is the problem... |
| 12:40.56 | fling | [TK]D-Fender: files are installed with package… |
| 12:40.56 | [TK]D-Fender | You don't rip out your security instead of fixed the user |
| 12:40.58 | WIMPy | You don't want the conf file to be world readable. |
| 12:41.44 | fling | [TK]D-Fender: I will just change gid, do not want my config files to be writeable by asterisk |
| 12:41.47 | [TK]D-Fender | that is a lethal error |
| 12:41.50 | fling | thanks! |
| 12:42.10 | [TK]D-Fender | <fling> [TK]D-Fender: I will just change gid, do not want my config files to be writeable by asterisk <- yes you do.... |
| 12:42.56 | fling | [TK]D-Fender: chmod 640 /etc/asterisk/* ; chown root:asterisk /etc/asterisk/* |
| 12:43.22 | fling | only asterisk will be able to only read… |
| 12:44.45 | fling | udp 0 0 0.0.0.0:5060 0.0.0.0:* 32173/asterisk |
| 12:46.01 | fling | WIMPy: [TK]D-Fender: it is listening now! thanks! It is a bug in gentoo package… I should fix it |
| 12:46.18 | [TK]D-Fender | <PROTECTED> |
| 12:47.06 | fling | I'm about to use sip only. Do I need any other config files? |
| 12:47.30 | [TK]D-Fender | ~book |
| 12:47.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:47.32 | [TK]D-Fender | ^^ |
| 12:47.34 | [TK]D-Fender | yes. plenty |
| 12:48.23 | Chainsaw | fling: So it's my fault then? |
| 12:48.35 | fling | Chainsaw: where, what? |
| 12:48.42 | Chainsaw | fling: You're blaming the Gentoo package. |
| 12:49.01 | fling | Chainsaw: I will make a patch on weekends |
| 12:49.28 | fling | Chainsaw: I want to make my setup first, I still did not made a single call :p |
| 12:49.41 | Chainsaw | fling: Or you run the config sections of the ebuild, specifically dedicated to this purpose? |
| 12:50.42 | fling | Chainsaw: just tried emerge --config asterisk |
| 12:51.04 | Chainsaw | fling: It sounds to me like you installed asterisk without USE=samples and then unpacking your own set. |
| 12:51.07 | fling | Chainsaw: 660 ? why? 640 should be |
| 12:51.27 | fling | Chainsaw: no, I have installed with USE=samples |
| 12:51.52 | fling | Chainsaw: and then removed some config files and useflag |
| 12:52.07 | Chainsaw | fling: So that loop with chown asterisk:asterisk $conffile only exists in my mind? |
| 12:53.19 | fling | Chainsaw: asterisk:asterisk? possible, because this is what I have after emerge --config > http://bpaste.net/show/33410/ ; root is the owner |
| 12:54.08 | Chainsaw | fling: Look at the ebuild code. The only way you can get it in this state is by installing without USE=samples. |
| 12:55.22 | fling | Chainsaw: yes, I removed the flag, because I did not wanted to install config files anymore… |
| 12:56.38 | Chainsaw | [TK]D-Fender: You are correct, in that the blaming of the package seems entirely misplaced. |
| 12:57.08 | [TK]D-Fender | Chainsaw, I never said anything about "packages" |
| 12:57.32 | [TK]D-Fender | Chainsaw, The problem is the files. I didn't give a rat's ass WHY they were the way they were :) |
| 12:57.59 | [TK]D-Fender | Chainsaw, .... silly ricer :p |
| 12:58.04 | elliot98 | remote fax server is trying to reinvite to t38, but asterisk responding with "Non Acceptable Here" |
| 12:58.21 | elliot98 | asterisk logs do show anything...why would this be happening? |
| 12:58.28 | elliot98 | *do not show anything |
| 12:58.32 | Chainsaw | [TK]D-Fender: You can use your RedHat Expensive Linux if you like. I really don't care. |
| 12:59.18 | fling | [TK]D-Fender: gentoo is for real men |
| 12:59.49 | [TK]D-Fender | Chainsaw, I like my CentOS Cheap, Slackware Suave, and Tarball Terrific, thank you ;) |
| 13:00.24 | [TK]D-Fender | fling, "gentoo is for" <- go loko at Google auto-complete |
| 13:00.26 | [TK]D-Fender | look* |
| 13:00.56 | [TK]D-Fender | http://funroll-loops.info/ |
| 13:01.03 | fling | [TK]D-Fender: lol! really funny :p |
| 13:01.08 | Chainsaw | fling: And in closing, 660 is because you might have web utilities writing to config files. You do not want to run your entire apache as asterisk. |
| 13:01.38 | fling | Chainsaw: oh! sorry then… did not knew this too |
| 13:05.04 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 13:07.24 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 13:08.14 | *** join/#asterisk GameGamer43 (users.5533@gateway/web/irccloud.com/x-ehglbbsqxirjqdgp) |
| 13:16.15 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-78-180.bois.qwest.net) |
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| 13:20.45 | fling | [Jul 4 20:20:06] ERROR[371] chan_sip.c: Peer '101' is trying to register, but not configured as host=dynamic |
| 13:20.58 | fling | but it is set to host=dynamic > http://bpaste.net/show/33411/ |
| 13:26.36 | fling | [Jul 4 20:20:01] ERROR[366] netsock2.c: getaddrinfo("dynamic ", "(null)", ...): Name or service not known |
| 13:26.40 | fling | [Jul 4 20:20:01] WARNING[366] acl.c: Unable to lookup 'dynamic ' |
| 13:28.08 | [TK]D-Fender | Ditch the comment |
| 13:29.13 | fling | [TK]D-Fender: hmm? |
| 13:33.53 | fling | [TK]D-Fender: thanks, bad how-to :p |
| 13:36.02 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:36.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:38.27 | fling | I can call myself! |
| 13:39.21 | [TK]D-Fender | Not much of an improvement from they voice's typical methods.... |
| 13:39.24 | [TK]D-Fender | the* |
| 13:44.19 | carrar | woot happy 4th |
| 13:46.24 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 13:49.23 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 13:49.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:50.45 | elliot98 | does t38 work if dialing a local extension? |
| 13:51.29 | elliot98 | such as Local/123@mydialplan |
| 13:53.28 | [TK]D-Fender | T38 is an end to end protocol. |
| 13:53.43 | [TK]D-Fender | SIP -> enpoint. If you bridge to another channel-type forget it |
| 13:53.47 | [TK]D-Fender | That's jsut silly |
| 13:54.47 | elliot98 | so even if it's local, it'll mess things up |
| 13:55.05 | elliot98 | local channel, that is |
| 14:03.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:06.40 | elliot98 | [TK]D-Fender: t38 doesn't kick in even when it's SIP <-> SIP |
| 14:07.14 | elliot98 | [TK]D-Fender: traces show Asterisk responds with a Not Accepted Here when the remote server tries to reinvite to t38 |
| 14:08.05 | [TK]D-Fender | If you say so.... |
| 14:09.25 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:18.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 14:22.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:30.14 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 15:05.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:09.32 | *** join/#asterisk SeRi (~wtf@c-98-200-177-50.hsd1.tx.comcast.net) |
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| 15:30.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:32.54 | *** join/#asterisk enki (~enki@kahuna.routed.com) |
| 15:32.57 | enki | hello |
| 15:35.05 | enki | I have a weird issue I was hoping someone could help me with. I have an asterisk (1.4.43) box on a public IP, and am trying to get a sip client on a cell phone to work on it. If I go over wifi (locally) it works fine, as it's on the LAN. But if I go over 3G, while the SIP traffic goes to the client's public (3g) address, the RTP traffic is going to the 10.x.x.x NAT'd IP. |
| 15:35.28 | enki | I have nat=yes in the respective sip.conf entry |
| 15:35.56 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 15:36.26 | enki | i confirmed via pcap that the SDP is showing the IN IP4 as 10.x.x.x. But I thought nat=yes was supposed to ignore that and use the inbound IP the traffic was coming from. |
| 15:37.57 | [TK]D-Fender | show us |
| 15:38.29 | enki | show what? |
| 15:38.38 | WIMPy | enki: That's the idea, yes. |
| 15:38.53 | WIMPy | Are you sure it matches the peer you think it should match? |
| 15:39.19 | WIMPy | Or do you have a global nat=yes? |
| 15:39.23 | enki | both |
| 15:40.42 | enki | http://pastebin.com/w2yG2mgH |
| 15:43.31 | enki | and all the sip traffic gets passed perfectly fine (invite, 100, 180, 200, BYE) |
| 15:43.55 | enki | i've also tried multiple clients, and all have the same issue, so it's definitely a server problem. |
| 15:43.57 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:44.28 | enki | also, I've tried the same setup (using the same sip.conf peer settings) on a different box running 1.4.23.2, and it works fine, which makes me even more confused. |
| 15:45.13 | *** part/#asterisk dijib (~dijib@bas10-kitchener06-1176139198.dsl.bell.ca) |
| 15:47.00 | [TK]D-Fender | enki, Show us the CALL |
| 15:47.03 | enki | and this is the invite if it helps: http://pastebin.com/ydEAQ3kH |
| 15:47.10 | *** join/#asterisk shadebob (~shadebob@41.248.242.85) |
| 15:47.22 | enki | do you want the entire thing or does the invite suffice? |
| 15:47.29 | [TK]D-Fender | Looks fine |
| 15:47.50 | enki | but here you can see the RTP going to the private IP: 11:06:47.784199 IP pool-71-180-124-149.tampfl.fios.verizon.net.13710 > 10.46.91.196.45994: UDP, length 172 |
| 15:48.01 | [TK]D-Fender | Show us the FULL call |
| 15:49.45 | enki | <PROTECTED> |
| 15:55.20 | enki | the only way I've been able to get remote calls working (if the remote end is behind NAT) is routing the traffic through a VPN tunnel. But obviously that isn't the best solution . |
| 15:56.06 | [TK]D-Fender | hrm... |
| 15:57.54 | *** join/#asterisk bdfoster_ (~bdfoster@unaffiliated/bdfoster) |
| 16:00.10 | enki | also, and I don't know if this matters, but the pcap shows the RTP traffic going from the client to the server, which would make me think I would at least get one-way audio, but I don't. No audio in either directions. |
| 16:01.58 | WIMPy | Ah |
| 16:02.32 | WIMPy | If you don't receive anything, Asterisk can't know where to send. |
| 16:02.41 | WIMPy | So you need to find out where it gets stuck. |
| 16:03.05 | enki | well I am receiving it (at least on the pcap): |
| 16:03.15 | enki | 11:06:47.685352 IP mobile-032-133-196-129.mycingular.net.45994 > pool-71-180-124-149.tampfl.fios.verizon.net.13710: UDP, length 172 |
| 16:03.16 | enki | 11:06:47.704320 IP pool-71-180-124-149.tampfl.fios.verizon.net.13710 > 10.46.91.196.45994: UDP, length 172 |
| 16:03.41 | enki | but asterisk doesn't seem to be recognizing it? Maybe because it's expecting to see the RTP from 10.46.91.196 |
| 16:04.39 | WIMPy | If you have nat=yes, it won't care. |
| 16:04.53 | enki | that's what I thought too |
| 16:08.01 | enki | BUT, when I do a pcap on the LAN side (going to the internal extension that answered), I am not seeing the RTP from the remote client, only the RTP from the internal phone. So Asterisk isn't even relaying the traffic it is receiving from the remote. |
| 16:11.12 | ChannelZ | Is * behind a firewall/NAT its self? |
| 16:11.41 | enki | no, on the edge with a public IP |
| 16:12.21 | [TK]D-Fender | enki, * 1.4 = EOL.... might see about fixing that... |
| 16:12.45 | enki | the box is running an iptables based FW, but it's allowing all SIP/RTP traffic through. And I have no issues with inbound/outbound calls in general, only when a remote client is NAT'd. |
| 16:13.03 | enki | [TK]D-Fender, i know. I plan on upgrading soon :D |
| 16:13.29 | enki | do you think it's a bug in asterisk? |
| 16:13.49 | WIMPy | Did you disable reinvites for your local phones? |
| 16:14.03 | enki | i can get it working on an earlier 1.4 version (different box), but can't figure out why it's not working on this one. |
| 16:14.49 | enki | WIMPy, yes, all extensions have canreinvite=no |
| 16:15.36 | ChannelZ | does your actual LAN have the same IP range 10.x.x.x? |
| 16:16.37 | enki | no, I use 172.30/21/20.0.0/24 internally. |
| 16:17.16 | enki | I also just tried calling an IVR destination, eliminating the local extension variable. Same result, no audio. |
| 16:20.15 | enki | i also confirmed in the asterisk 'sip show peer' that NAT : Always (for that remote extension) |
| 16:29.24 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 16:32.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:32.48 | *** join/#asterisk GGD (~deberle@pool-173-72-204-101.clppva.fios.verizon.net) |
| 16:37.08 | *** join/#asterisk din3sh (din3sh@41.136.85.117) |
| 16:39.06 | enki | so any ideas? |
| 16:46.38 | enki | hrmmm, new update. So the Bria softphone client seems to have a function to discover/use it's global IP for NAT transversal. When I enable this, the "Reg. Contact" changes to show the public IP, and then audio works! |
| 16:47.07 | enki | so it's definitely an issue with Asterisk not being able to properly handle a NAT'd peer. |
| 16:49.17 | [TK]D-Fender | Apparently |
| 16:49.23 | [TK]D-Fender | that one version anyway |
| 16:49.40 | enki | :( |
| 16:50.05 | enki | i kind of wanted to use the built in SIP client in android ICS |
| 16:50.16 | enki | which doesn't support global IP discovery |
| 16:54.03 | *** join/#asterisk PaybackTony (473b84ca@gateway/web/freenode/ip.71.59.132.202) |
| 16:58.19 | enki | how much dialplan/configuration changes are necessary if upgrading 1.4 to 1.8? |
| 17:00.04 | ectospasm | enki: see all UPGRADES.txt in 1.8 source |
| 17:04.46 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 17:07.07 | PaybackTony | I have a question maybe someone can help me with. I'm attempting to get a simple PHP agi script working. I had one working previously, but that was with a binary installation. Once I switched to a version from source, my php agi scripts seemed to have stopped working. I tested with the test-agi.agi script and it works. By not working I mean, nothing special is spit out in the agi debug and the script doesn't have any errors whe |
| 17:07.18 | PaybackTony | http://pastebin.com/ARLxBA5N <- the agi |
| 17:09.00 | PaybackTony | It gets to where the normal output would be in the agi debug log via cli and then the call ends and nothing else is spit out (not even an exit code) |
| 17:09.02 | PaybackTony | http://pastebin.com/T8PiJKdw |
| 17:10.51 | [TK]D-Fender | PaybackTony, You're already not following the basics for PHP AGI's.... go read some more samples |
| 17:12.23 | PaybackTony | The paste was just a very stripped out version of a sample on voip-info |
| 17:12.57 | [TK]D-Fender | link it |
| 17:14.06 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 17:19.28 | PaybackTony | Sure, let me dig it up again. One thing to note is that in many of the examples the header is slightly different (with -q as php parameter). Which gave me a "Extension '/var/lib/asterisk/agi-bin/agi.php' not present." message in the agi debugger. I removed the -q in the binary build and the script worked. No longer the case once compiled from source. |
| 17:28.35 | PaybackTony | http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
| 17:32.03 | jeffspeff | I've been trying to read up on realtime mysql configs. Something just doesn't sound right... I read that realtime SIP can't do voicemail notifications and NAT keep-alives... About a year ago I implemented realtime SIP through Active Directory LDAP, if i remember correctly, I indeed had NAT keep-alives... can anybody verify the absence of voicemail notifications and NAT keep alives in realtime sip? |
| 17:33.44 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:50.57 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 17:58.20 | elliot98 | how does one use a previous menuconfig setup? |
| 18:01.13 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:06.59 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:10.41 | elliot98 | how does one install the t38 spandsp fax functionality? |
| 18:11.59 | *** join/#asterisk danfromuk (~IceChat77@2.27.11.133) |
| 18:12.14 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:12.53 | danfromuk | Hi, Ive just upgraded to 1.8 and fail2ban doesnt seem to be working with the new version. |
| 18:13.02 | danfromuk | Has anyone set up fail2ban with 1.8? |
| 18:13.43 | danfromuk | There is a old user that keeps trying to access the asterisk server but fail2ban should be blocking them after a few attempts. However its not. |
| 18:17.11 | [TK]D-Fender | And if logging even enabled? Didi you loko at the rules you're using? |
| 18:17.17 | [TK]D-Fender | is* |
| 18:19.36 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 18:21.20 | eduzimrs | Hey guys, do u have best practices to extend the range of branches? im using for internal calls (exten => _[134]XX,1,Dial(SIP/${EXTEN})) and its reaching the last free number |
| 18:21.46 | [TK]D-Fender | eduzimrs, "branches"? "range"? |
| 18:22.09 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 18:22.13 | [TK]D-Fender | eduzimrs, Please rephrase your question |
| 18:22.17 | eduzimrs | sry |
| 18:24.20 | eduzimrs | [TK]D-Fender, i dont know how do u treat this, but i`d like to extend 1 number more to be able to dial, cause i have no free numbers to dial anymore, understand? |
| 18:25.20 | [TK]D-Fender | No. |
| 18:25.34 | [TK]D-Fender | What is "1 number more"? |
| 18:26.02 | [TK]D-Fender | Patterns aer patterns. If you want more than that range then the rules are all documented |
| 18:26.44 | danfromuk | [TK]D-Fender, logging is enabled and i can see the No Match Peer Found message is being written in the asterisk logs. |
| 18:27.01 | [TK]D-Fender | danfromuk, Go look at your fail2ban rules then |
| 18:28.31 | eduzimrs | [TK]D-Fender, tecnically speaking i do not know how to do this, maybe adding 1 in front of like: _1[134]XX |
| 18:28.37 | *** join/#asterisk hudony (~chatzilla@modemcable078.217-70-69.static.videotron.ca) |
| 18:28.50 | [TK]D-Fender | eduzimrs, that makes for a 5 digit pattern |
| 18:28.58 | [TK]D-Fender | 4 rather |
| 18:29.08 | danfromuk | [TK]D-Fender: It all looks ok. Heres the asterisk.conf from fail2ban |
| 18:29.09 | danfromuk | http://pastebin.com/46L0Zcbj |
| 18:29.13 | hudony | Hi there, I tried so search and cannot find anything explicit about the difference between the timeout in the Dial() application and the RINGTIMER global variable in extensions.conf |
| 18:29.36 | hudony | I fully understand the timeout in Dial() but not sure for RINGTIMER |
| 18:29.44 | danfromuk | Ive not made any changes. its taken from an example on voip-info. Worked fine with 1.8. |
| 18:29.54 | danfromuk | It could be that the polling method isnt working. |
| 18:30.00 | danfromuk | but ssh f2b is working fine |
| 18:30.56 | [TK]D-Fender | danfromuk, show the actual log entries |
| 18:31.08 | danfromuk | From asterisk or f2b? |
| 18:31.41 | [TK]D-Fender | YES |
| 18:31.57 | danfromuk | asterisk logs: http://pastebin.com/KFewxMzy |
| 18:33.42 | [TK]D-Fender | hudony, Where do you see this channel variable documented as being tied to some specific functionality? |
| 18:34.26 | hudony | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Globals |
| 18:34.52 | hudony | oh |
| 18:35.02 | hudony | Don't tell me |
| 18:35.24 | hudony | This is not a system variable... |
| 18:35.57 | hudony | Well...That actually explains a lot I guess :D |
| 18:36.42 | [TK]D-Fender | danfromuk, Wonder if the timestamp in front is interfering |
| 18:38.21 | *** join/#asterisk wonderworld (~ww@dsdf-4db53dcb.pool.mediaWays.net) |
| 18:40.28 | *** join/#asterisk navaismo (~navaismo@189.144.194.18) |
| 18:41.54 | danfromuk | Maybe, although it worked in the past with 1.4 with the timestamp |
| 18:42.24 | *** join/#asterisk Kyosh (~whoa@pool-74-108-131-13.nycmny.fios.verizon.net) |
| 18:45.10 | danfromuk | I believe the timestamp is required to determine how often the attempts are being made. |
| 18:47.46 | elliot98 | is there a way to know which version of spandsp is used for the res_fax_spandsp? |
| 18:49.59 | *** join/#asterisk neurosys (~neurosys@c-98-254-122-167.hsd1.fl.comcast.net) |
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| 19:32.11 | *** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 19:35.18 | mcf3782 | Can someone tell me if this is syntactically correct? The desired effect is "If the length of the string MEETME_ROOMNUM is equal to 4, go to context ext-meetme,1" |
| 19:35.20 | mcf3782 | exten => 1,n,GotoIf($[${LEN(${MEETME_ROOMNUM})} = 4 ]?ext-meetme,1) |
| 19:39.03 | [TK]D-Fender | Looks about right, though I'd get rid of the extra whitespace |
| 19:41.41 | mcf3782 | OK. Does the extra white space possibly effect the proper functioning of the line, or are you suggesting to get rid of it just because it's extraneous? |
| 19:42.48 | [TK]D-Fender | both |
| 19:43.14 | mcf3782 | Hmm… ok. Maybe that's why it's not working then. :) |
| 19:43.55 | mcf3782 | So this instead: |
| 19:43.56 | mcf3782 | exten => 1,n,GotoIf($[${LEN(${MEETME_ROOMNUM})}=4]?ext-meetme,1) |
| 19:44.24 | [TK]D-Fender | you might want to consider trying it... |
| 19:45.21 | mcf3782 | I am. :) I'm just clarifying what "white space" you were talking about. :) |
| 19:48.35 | [TK]D-Fender | you already got it all |
| 19:49.39 | redax | hi |
| 19:50.12 | mcf3782 | It doesn't seem to be taking the 'ext-meetme,1' as a switch to a different context. It looks like it's trying to….?….find a label in the current context….?…. |
| 19:50.14 | mcf3782 | <PROTECTED> |
| 19:51.03 | redax | is the usual GSM->SIP gateways can handle the 2nd call in case the GSM line is busy? |
| 19:51.24 | [TK]D-Fender | mfcYou aren't showing us your actual dialplan. |
| 19:51.37 | [TK]D-Fender | mcf3782, So we have no idea what else whe should expect |
| 19:51.46 | mcf3782 | ok. I can fix that. :) |
| 19:51.54 | [TK]D-Fender | mcf3782, Who says the GotoIf is bad... if we don't know the TARGET? |
| 19:51.55 | redax | I imagine it like, it should leave a missed call in my phone... don't you know the GSM->SIP adapters support this kind of feature? |
| 19:54.14 | danfromuk | [TK]D-Fender: Resolved the fail2ban issue. <HOST> does not include the port number which I thought it did. Maybe in 1.4, the port number wasn't included in the logs. And in 1.8 it is included in the logs. |
| 19:54.34 | jeffspeff | when using setvar in realtime mysql, how do i set multiple variables? would it be table (setvar) -> var1=value1, var2=value2 ? |
| 19:55.21 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:56.22 | wdoekes | jeffspeff: semi-colon separated.. like the codec allow lists |
| 19:56.57 | WIMPy | redax: That obviousely depends on the device. |
| 19:57.04 | mcf3782 | [TK]D-Fender: http://pastebin.com/5LXzm5si |
| 19:57.18 | jeffspeff | wdoekes, should there be a space after the semi-colon before the next variable? |
| 19:57.59 | [TK]D-Fender | mcf3782, Now try a real one where we can see ALL of the context names |
| 19:58.35 | [TK]D-Fender | mcf3782, -- Sent into invalid extension 'ext-meetme' in context 'gc-conf-bridge' on SIP/501-0000004c |
| 19:58.42 | redax | WIMPy: have you seen any GSM-SIP gw which supports this kind of feature? |
| 19:58.44 | [TK]D-Fender | mcf3782, and as we see here, we HAVE the context name |
| 19:58.59 | [TK]D-Fender | and I do NOT see that context tage in there.. and clearly no exten to match. |
| 19:59.10 | WIMPy | redax: I can't see how SIP could do it. |
| 19:59.12 | [TK]D-Fender | mcf3782, You aren't looking at your TARGET |
| 19:59.53 | [TK]D-Fender | mcf3782, extension very clearly does not exist |
| 20:00.54 | mcf3782 | ….does not exist…… Then I'm way off in the weeds somehow. |
| 20:01.04 | jeffspeff | lol |
| 20:01.17 | mcf3782 | ok. I'll go back to staring at it and scratching my head. |
| 20:01.25 | [TK]D-Fender | mcf3782, What line # in that PB do you think it should successfully go to? |
| 20:02.12 | [TK]D-Fender | <PROTECTED> |
| 20:03.17 | mcf3782 | What I *thought* I was sending it to, is line…..15 "[ext-meetme]". I just want to transfer control over to that context in the dial plan if the number of digits read in line….5…is 4 digits. |
| 20:03.38 | [TK]D-Fender | exten => 1,n,GotoIf($[${LEN(${MEETME_ROOMNUM})}=4]?ext-meetme,1) |
| 20:03.50 | [TK]D-Fender | ext-meetme,1 <--- exten, priority |
| 20:04.09 | mcf3782 | So, I can't specify transfering to a context? |
| 20:04.21 | [TK]D-Fender | mcf3782, you have to specify ALL THREE |
| 20:04.26 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.40.134) |
| 20:04.36 | jeffspeff | should be more like -> ext-meetme,s,1 |
| 20:04.39 | [TK]D-Fender | mcf3782, Context,Exten,Priority |
| 20:04.59 | mcf3782 | OH! |
| 20:05.13 | [TK]D-Fender | You don't get to skip a level of precision |
| 20:05.26 | [TK]D-Fender | P, EP, CEP |
| 20:05.48 | [TK]D-Fender | You always need through to the least. |
| 20:06.01 | [TK]D-Fender | lowest* |
| 20:06.31 | [TK]D-Fender | Priority required, exten optional. If you provide exten, context is optional. |
| 20:07.23 | mcf3782 | OK. Clearly I read the docs to GotoIf wrong. I didn't see the requirement for Exen,Priority. I'll re-read it again. |
| 20:09.06 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 20:13.30 | elliot98 | in asterisk 1.8, can variables be put in quotes as in: Set("AST_CR"="5")? |
| 20:13.36 | redax | WIMPy: like a second call, just it hangs up after a few seconds :) |
| 20:15.54 | WIMPy | redax: Maybe you should disable CW. |
| 20:17.32 | [TK]D-Fender | elliot98, No |
| 20:18.46 | redax | WIMPy: actually I need some feature like the call waiting on GSM gateways |
| 20:19.21 | redax | would be good to know who called me, meanwhile I were busy... |
| 20:19.41 | *** join/#asterisk gusto (~gusto@ppp-62-216-206-44.dynamic.mnet-online.de) |
| 20:19.41 | gusto | here i am |
| 20:19.51 | gusto | with a brand new DSL connection :-) |
| 20:20.13 | gusto | whaha |
| 20:20.45 | gusto | somehow that **** fritzbox is blocking 5060 port |
| 20:21.23 | gusto | that's really bad because now i have to switch to 5061 for outside connectivity, or to IAX2 ... i am going to try IAX2 first |
| 20:22.09 | gusto | plan b.) is of course putting the old modem on it, and that fritzbox to the box ;-) |
| 20:23.14 | *** part/#asterisk GGD (~deberle@pool-173-72-204-101.clppva.fios.verizon.net) |
| 20:25.21 | WIMPy | redax: As I said: I can't see how a SIP GW could do that. |
| 20:28.47 | redax | WIMPy: ok, thank you |
| 20:29.00 | mcf3782 | Ah! |
| 20:29.07 | mcf3782 | What I really wanted was: |
| 20:29.11 | mcf3782 | exten => 1,n,GotoIf($[${LEN(${MEETME_ROOMNUM})}=4]?ext-meetme,${MEETME_ROOMNUM},1) |
| 20:29.50 | mcf3782 | Now that I see it, it makes perfect sense. It just didn't from reading the docs and the examples. |
| 20:29.55 | mcf3782 | *sigh* |
| 20:40.36 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 20:50.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 20:58.42 | mcf3782 | [TK]D-Fender: I figured it out. :) Thanks for the pointers! |
| 20:59.23 | *** join/#asterisk danfromuk (~IceChat77@2.27.11.133) |
| 20:59.36 | gusto | somehow i have a problem with that, my IAX does not authenticate |
| 20:59.44 | gusto | i suspect that's because he has no STUN |
| 21:01.02 | WIMPy | That doesn't make sense. |
| 21:01.43 | *** join/#asterisk wonderworld (~ww@dsdf-4db53dcb.pool.mediaWays.net) |
| 21:03.06 | gusto | well, if it sends the answer to the wrong IP? |
| 21:04.20 | WIMPy | IAX doesn't use RTP, so there's no SDP, either. |
| 21:04.34 | WIMPy | Anything alse would be a real networking issue. |
| 21:04.36 | gusto | i am stuck on registration send, and no answer, but it's ok, because when i did write it wrongly, it came with a reject |
| 21:06.07 | [TK]D-Fender | IAX doesn't care about stun |
| 21:08.01 | danfromuk | Anyone know why Qpid on an asterisk server would be running at 40 to 50%? No idea what Qpid is used for. |
| 21:08.49 | *** join/#asterisk wonderworld (~ww@dsdf-4db53dcb.pool.mediaWays.net) |
| 21:10.30 | gusto | ok |
| 21:10.46 | gusto | so i only need STUN when my IP changes every day, right? |
| 21:10.52 | [TK]D-Fender | no |
| 21:11.02 | gusto | i mean for the registrys |
| 21:11.10 | [TK]D-Fender | * simply needs to know its IP. |
| 21:11.15 | gusto | or do they renew it? |
| 21:11.26 | [TK]D-Fender | dyndns works just fine as do several other discovery methods |
| 21:11.47 | gusto | no, i mean from a VoIP provider |
| 21:11.55 | [TK]D-Fender | STUN isn't magic. Please avoid treating it as anything necessary |
| 21:12.02 | mcf3782 | danfromuk: I think Qpid is an Apache module…. http://qpid.apache.org/ |
| 21:12.04 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 21:12.04 | gusto | a VoIP provider can not know what IP i have, or dyndns |
| 21:12.35 | gusto | but however, i can set reregister at say 5 minutes and then he should see that my ip has changed, right? |
| 21:12.36 | [TK]D-Fender | gusto: You clearly have no idea what DynDNS is... |
| 21:12.47 | danfromuk | mcf3782: is there a problem to turn it off if i'm not using apache? |
| 21:13.18 | [TK]D-Fender | gusto: The point is to keep * up to date on what to fill the Contact: header with |
| 21:13.39 | [TK]D-Fender | gusto: "externhost" + DynDNS service |
| 21:14.04 | mcf3782 | I wouldn't think so.. Do you know if Apache is running on the box? It's possible that what's on your box is not the same as the Apache module. |
| 21:14.54 | *** join/#asterisk shadebob (~shadebob@41.141.239.34) |
| 21:14.55 | danfromuk | The server is just running asterisk. |
| 21:15.00 | danfromuk | Mysql is on another server. |
| 21:15.01 | *** join/#asterisk abissom (~user@unaffiliated/sicelo) |
| 21:15.17 | danfromuk | The server is also running samba. But apart from that, nothing else. |
| 21:16.09 | mcf3782 | If you're sure that Apache isn't installed and running, then your "Qpid" must be something different.. |
| 21:17.51 | gusto | [TK]D-Fender: well, it seems to have worked without |
| 21:18.13 | danfromuk | I've stopped it and will monitor the server to see if anything fails. |
| 21:18.23 | gusto | [TK]D-Fender: now, i do not know what, but i did not do anything, so it must be the password, that it was not set on the side of the provider yet, because that account was newly created |
| 21:19.14 | gusto | ok, i have registered state now with refresh of 60 seconds, so when my IP changes, he gets to know it within 60 seconds maximum, right? |
| 21:19.32 | [TK]D-Fender | gusto: "Must be"? You show no evidence |
| 21:19.48 | [TK]D-Fender | gusto: You are making blind guesses |
| 21:19.54 | gusto | [TK]D-Fender: well, the evidence is that i did not do anything |
| 21:20.11 | gusto | [TK]D-Fender: well, its not that i could look that up on the providers's side |
| 21:20.13 | mcf3782 | danfromuk: Sounds like a reasonable plan. From a security point of view, if there is something running on your box that you didn't install and don't know what it does; then it's probably a good idea to figure out what's starting it and who installed it. :) |
| 21:20.16 | [TK]D-Fender | No, that isn't evidence", that is merely a lack of action on your personal part |
| 21:20.43 | gusto | [TK]D-Fender: shall be |
| 21:21.07 | gusto | however, we have registered, that is a good step forward |
| 21:22.31 | danfromuk | mcf3782: just checked our other servers and qpidd is running on the other servers too. Must have been included in the base centos install. |
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| 21:23.33 | abissom | what does qpidd do? |
| 21:23.48 | danfromuk | No idea. |
| 21:24.03 | abissom | google? |
| 21:24.08 | danfromuk | tried that |
| 21:24.12 | mcf3782 | If it's not a service/application/function that you need; then turning it off is probably a good thing. No point in wasting resources with something that you're not using. |
| 21:24.17 | danfromuk | the answer was a bit over my head |
| 21:25.09 | abissom | hmm, interesting. i agree with mcf3782 though. |
| 21:25.32 | danfromuk | Doing that. I'll keep an eye on it and see whats happening |
| 21:27.33 | mcf3782 | danfromuk: try running "yum search qpid" as root on your box. See if it gives you any information |
| 21:28.18 | danfromuk | <PROTECTED> |
| 21:28.22 | abissom | there actually seems to be a 'man qpidd' |
| 21:28.34 | danfromuk | The servers are virtualised so may be related |
| 21:29.49 | gusto | what is a timing interface? |
| 21:32.00 | mcf3782 | It's *probably* not related to Asterisk, but it could be important to how your VMs communicate with the VM controller or with other VMs. If you've turned it off, keep an eye on the log file in /var/log; for signs of possible problems. |
| 21:32.22 | mcf3782 | *files |
| 21:32.53 | danfromuk | i'll check with the linuxkvm channel to see if they know where its come from and if its important |
| 21:33.05 | danfromuk | thanks for your advice |
| 21:33.14 | mcf3782 | np. good luck. |
| 21:34.33 | mcf3782 | oops. Building-wide power outage ( |
| 21:34.42 | dym | evenings! |
| 21:34.55 | mcf3782 | scheduled) to start in 20 minutes. Time to shut down and get out of here. |
| 21:35.14 | dym | hurry (: |
| 21:35.34 | mcf3782 | yep |
| 21:35.39 | mcf3782 | *poof* |
| 21:35.41 | *** part/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
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| 22:38.53 | gusto | tss ... |
| 22:39.03 | gusto | you will all laugh now what happend to me |
| 22:40.20 | gusto | now, independently from that IAX what i was trying i also wanted to get the new telephone to work then |
| 22:40.52 | gusto | well, but i have no sip login data and so i thaught, what if it's an analogue telephone line? ... and it is ... :-D LOL |
| 23:01.00 | gusto | i have an analogue phone ;-) |
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| 23:27.08 | TechSmurf | are there any preferrable extension layouts for *now? |
| 23:27.52 | *** join/#asterisk RobZombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
| 23:29.43 | RobZombie | Anyone here? |
| 23:30.10 | [TK]D-Fender | Nope |
| 23:30.16 | RobZombie | lol |
| 23:30.36 | RobZombie | Hppy 4th of July, I was trying to see if someone had a solution for a problem I'm having. |
| 23:32.16 | chuckf | no one knows the problem you're having, so no one here has a solution to offer |
| 23:32.39 | RobZombie | Basically, I am trying to install AsterskNOW (Current Release) onto a netbook, install via a usb. I Used Unetbootin to iso to usb, worked fine for ubuntu but, with AsreriskNOW it just doesnt work, I have wnt all over documents on the web and see tons have this problem and no solution, and the problem dates back couple years. |
| 23:33.06 | TechSmurf | It just doesn't work. |
| 23:33.29 | RobZombie | I was afraid of that, so even if I get a usb cdrom it want work either, for my netbook, right? |
| 23:33.32 | chuckf | RobZombie: try using the fedora liveusb creator tool |
| 23:33.38 | TechSmurf | no, usb cd works fine |
| 23:33.45 | TechSmurf | just not flash drive |
| 23:34.23 | TechSmurf | CentOS throws a shit fit because the installer is looking exclusively for a cdrom device to install from |
| 23:34.43 | RobZombie | Okay, I had found a site where a guy showed doing it via usb on a video talked about 3 edits and then certain way the usb was edited, tried it, I was thinking, its just not gonna work, thanks for straight answer. |
| 23:35.02 | TechSmurf | well, I'm sure you could fix it given enough effort |
| 23:35.27 | RobZombie | I feel foolish I tried like two dozen methods.. Formating all sorts of crap. |
| 23:35.31 | RobZombie | So 1 other question |
| 23:36.01 | RobZombie | Since Asterisk now from usb is nor an option. Currently I'm using elastix. |
| 23:36.25 | RobZombie | I only went with elastix because when I installed 1.8 only of asterisk, the meetme stuff and other things werent there |
| 23:36.31 | RobZombie | elastix was a turn key solution |
| 23:36.43 | RobZombie | now im trying to add a phpmyadmin for a gui database solution |
| 23:36.47 | RobZombie | but it breaks the pbx |
| 23:37.22 | RobZombie | Anyone have any idea of a Ready to Run, works with usb install, that has all asterisk needs like elastix, but has a gui database as well? |
| 23:37.55 | TechSmurf | breaks the pbx as in calls don't go through, or breaks the web front end? |
| 23:38.36 | RobZombie | Break the elastix interface gui, which i dont use anyways except to backup and restore, and the call routes held in elastix database get all jacked up and cant be repaired |
| 23:38.47 | TechSmurf | oh my. |
| 23:38.58 | TechSmurf | shrugs |
| 23:39.15 | TechSmurf | outside the scope of my needs. |
| 23:40.00 | RobZombie | So basically, im look for a ready to run, can be installed via usb, that has everything asterisk needs when install is complete, like the meetme app so etc and has gui database. |
| 23:42.43 | TechSmurf | I think you may be asking for a bit much in one shot |
| 23:42.50 | RobZombie | I have only ever used asterisk 1.2 and a custom-core version of 1.2 at that, and now elastix, so im all noobalar. Oneof my friend employees got the phpymadmin working with elastix, but he has been busy working, on projects so has had no time, he says its a bit installed and requires a bit of understanding of the elastix use db that I dont have and he cant help me right now and I should try a different ready to run solution. |
| 23:43.08 | TechSmurf | Installing *now from usb-cd and adding myphpadmin will likely be the closest you get |
| 23:43.12 | RobZombie | Well I have a fully working system i was able to script together on my own, with tons of callers. |
| 23:43.30 | RobZombie | now i need a gui database so i can make some more advance functions and dialplan options. |
| 23:43.45 | RobZombie | Gotcha |
| 23:43.56 | RobZombie | , Appreciate the honesty. |
| 23:44.06 | TechSmurf | There are few good free asterisk distros |
| 23:44.41 | TechSmurf | I don't believe *now relies on a database for any of its functionality, so adding that shouldn't break a damn thing |
| 23:45.06 | RobZombie | Right thats why it was my first choice after reading, just dont have a usb-cd right at moment. |
| 23:45.18 | RobZombie | So i';ll need to address that I gues before i can move on |
| 23:45.26 | TechSmurf | spare sata cd hanging out of the box by its cables? :P |
| 23:45.39 | RobZombie | Yeah |
| 23:45.49 | TechSmurf | You only need it for 30 mins |
| 23:45.51 | RobZombie | I have several broke down pcs |
| 23:46.52 | RobZombie | I also have one of those master cloner usb's, adpaters somewhere for cloning drives, I wonder if I could get a cd on it? |
| 23:47.11 | RobZombie | hd's pop right on the adapter |
| 23:47.42 | WIMPy | What about putting the image somewhere on the network? |
| 23:48.05 | RobZombie | I have considered that, but I have never done a network boot before not sure how it works |
| 23:48.23 | TechSmurf | The installer didn't seem friendly with that option either |
| 23:48.33 | RobZombie | its an option on the f12 of the netbook to boot from network resource |
| 23:48.37 | WIMPy | Bad installer. |
| 23:48.47 | TechSmurf | yes, terrible installer |
| 23:49.16 | RobZombie | I tried like alot of methods people said sure work all failed, right now i have desk version of ubuntu on the netbook |
| 23:49.22 | TechSmurf | Actually just an hour or so ago I confused it terribly trying to stack raid/lvm... it tried to pvcreate before mdadm --create and borked the whole install |
| 23:49.42 | RobZombie | I wish I knew that custom versionof ubuntu running around that has a ready to run asterisk pre-config on it. |
| 23:49.47 | RobZombie | Cant find it either, i have tried |
| 23:49.52 | TechSmurf | had to boot up rescue and make the arrays first |
| 23:50.44 | WIMPy | I heard on the LUG camp that RH had a version of MD that was incompatible to standard kernels once. Can't beat that. |
| 23:51.24 | TechSmurf | but *now seems rather nice once the cent installer is out of the way |
| 23:52.18 | RobZombie | Alright how about, anyone know a version or distro of asterisk, 1.6 + if possible, that does a ready to run solution, what I mean is you install linux manually, then install it, because my only problem with 1.8 only install, everything worked but meetme.app.so wasnt present so I couldnt build conferences, when I yum'd the meet app, exact version it called for, would work, it showed as installed and functional but wouldnt work |
| 23:52.55 | RobZombie | and then i found articals about it after the fact too, so i zapped it and went with elastix it gave me all my asterisk requirements in a single 15 minute install to build off, but hard to install a gui db without killing the elastix |
| 23:53.02 | WIMPy | meetme requires dahdi. |
| 23:53.15 | RobZombie | Yea @ wimpy |
| 23:53.20 | RobZombie | I put it all in. |
| 23:53.23 | RobZombie | still didnt work |
| 23:53.41 | WIMPy | You can use ConfBridge instead. But the 1.8 version wan't that great. The versio on 10 outperforms meetme, however. |
| 23:54.02 | RobZombie | I was going to try v10 but i was afraid of it |
| 23:54.05 | RobZombie | like buggy? |
| 23:54.31 | WIMPy | Anything from 1.8 upwards should be fine. |
| 23:55.23 | RobZombie | So you dont think 10 should be waited on for a noob like me? since its not stable yet? I mean im a beast int he dp wiritng things, but linux, and general install and setup config, im like, useless. |
| 23:55.34 | RobZombie | I have always worked within already configured systems. |
| 23:55.46 | WIMPy | 11 isn't stable, yet. |
| 23:55.53 | RobZombie | 10 is stable now? |
| 23:55.57 | RobZombie | huh? |
| 23:56.15 | WIMPy | What makes you think it might not be? |
| 23:56.31 | RobZombie | The asterisk/digium website doesnt have a stable downlaod |
| 23:56.33 | RobZombie | is why i thought that |
| 23:56.59 | RobZombie | Highest stable there was 1.8 |
| 23:57.18 | WIMPy | I never look at packages, but I'd certainli prefer 10 over anything <1.8. |
| 23:57.43 | WIMPy | And the new ConfBridge is a good point for 10. |
| 23:57.49 | RobZombie | So does the 10 package come ready to run? |
| 23:58.12 | RobZombie | Cause im noobalar city here mate, literally. |
| 23:58.43 | WIMPy | I don't know. I have only ever used the source. |
| 23:58.58 | RobZombie | You wanta crazy dp that exercises tons of params and func and does great things, I can do, setup and install and config, im like a dead horse, useless. |