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| 02:12.51 | tompaw | That's quiet. |
| 02:20.29 | wyoung | It is |
| 02:20.40 | wyoung | but we just ruined it |
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| 04:35.57 | snadge | From: "301-09541696" <sip:301-09541696@202.xx.xx.xx>;tag=56c074da8cc025dbo5 |
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| 04:36.58 | snadge | not strictly an asterisk question but... if i send a call to a cisco device registered as an extension.. when you miss the call it shows up as "301-09541696" 30109541696 .. so it appears to strip the - from the sip: field |
| 04:37.30 | snadge | so when you hit redial.. its an invalid number.. its a stupid quirk of thirdlane.. 301 is the extension number, and 09541696 is the tenant number |
| 04:38.16 | snadge | piling through cisco docs isnt helping me much.. but i want it to dial the quoted number.. not sure what that field is called.. name ? |
| 04:38.30 | snadge | .. or not strip the - from the number |
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| 06:04.44 | robink_ | is not robmal |
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| 06:14.36 | dhawani | hi, can anyone explain what is "requested media update control 26"? |
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| 08:03.16 | mrfrenzy | Hi! I've been reading about pattern matching but cannot get it to work, could you please give some hints? |
| 08:03.47 | mrfrenzy | I have an inbound route that sometimes sends DID "0381556000" and sometimes it sends "u0381556000" |
| 08:04.08 | mrfrenzy | how could I make a route that matches both? |
| 08:17.01 | ChannelZ | The unambiguous way is to just make two extensions and make one jump to the other.. like exten => u0381556000,1,Goto(0381556000,1) and make your main 0381556000 do all the normal work |
| 08:20.58 | ChannelZ | And actually I think that'd be the only way really, asterisk doesn't really have a 'zero or more' matches metacharacter like regex. It does but only for ends of patterns. |
| 08:21.55 | mrfrenzy | that is what I'm just figuring out :( using . or ! in the beginning of the string is no good |
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| 09:39.57 | hirogen1 | hi |
| 09:40.03 | hirogen1 | anyone work for BT aka british telecom in uk ? |
| 09:40.16 | hirogen1 | urgently need a broadband business package re-enabled cos bt cut it off by mistake and its my presidents |
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| 10:39.51 | Milos | I know it's possible to use ChanSpy to listen and talk on an existing call, but if the original initiator exits the call, I am assuming the call will be hung up accordingly even if someone is still listening (which makes sense.) How can I make it so that someone else can join an existing call easily and if the original initiator hangs up the call remains active with the other person? |
| 10:59.44 | markusl | Milos: maybe you can transfer all three channels into a conference as soon as a 3rd party enters. |
| 11:00.08 | markusl | Milos: so you have a three way conference, if one party hangs up the other two can still talk |
| 11:00.27 | markusl | I don't know how to do that exactly right now |
| 11:00.47 | Milos | Yeah that's what I thought I'd have to do. With a conventional telephone you can just pick up and the other party can hang up, and with ChanSpy you just pick up and dial one number. How much would the complexity increase with what you've mentioned? |
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| 11:36.27 | markusl | i don't know. might be possible with a bit of dialplan |
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| 12:37.13 | juned | Where exactly i can get CAll end time in dialplan ? |
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| 12:37.53 | juned | <PROTECTED> |
| 12:38.23 | juned | i am getting null value |
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| 12:53.34 | aiksa[LV] | juned - I would assume call has not ended yet |
| 12:53.52 | juned | aiksa[LV]: that's true |
| 12:54.30 | juned | So where exactly I should get this value ? |
| 12:54.40 | aiksa[LV] | therefore - if the call is not ended there is no ended_time value yet. |
| 12:55.22 | juned | Ohh okay |
| 12:55.39 | aiksa[LV] | if I need that data - I usually resort to AMI and subscribing to CDR event |
| 12:55.48 | juned | so is there any other way to get this value within a dialplan |
| 12:56.02 | aiksa[LV] | h extension could work |
| 12:56.11 | aiksa[LV] | but I am not sure |
| 12:56.24 | juned | aiksa[LV]: no problem I ll give a try :) |
| 12:56.34 | juned | Thank you.. |
| 12:57.43 | juned | aiksa[LV]: I've one more question |
| 12:58.09 | juned | can i log custom fileds in cdr log (Master.csv) ? |
| 12:58.20 | juned | fields* |
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| 13:06.15 | aiksa[LV] | cdr_custom.conf |
| 13:06.20 | aiksa[LV] | take a look at this |
| 13:06.26 | aiksa[LV] | it should help |
| 13:08.45 | juned | aiksa[LV]: yeah I looked into that but seems some issue with that |
| 13:09.02 | juned | even default configuration is not working for me |
| 13:09.12 | juned | giving error like this : |
| 13:09.12 | juned | WARNING[9571]: config.c:1803 process_text_line: parse error: No category context for line 13 of /etc/asterisk/cdr_custom.conf |
| 13:09.12 | juned | [Aug 27 18:38:24] ERROR[9571]: cdr_custom.c:99 load_config: Unable to load cdr_custom.conf. Not logging custom CSV CDRs. |
| 13:09.41 | aiksa[LV] | and whats on line 13? |
| 13:10.02 | juned | Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})}, |
| 13:10.21 | juned | its a default one i've just uncommented it |
| 13:12.13 | aiksa[LV] | do you have [Mappings] uncommented? |
| 13:12.25 | aiksa[LV] | [mappings] |
| 13:12.43 | juned | Ohh Man.......it is commented |
| 13:12.48 | juned | Lolz :) |
| 13:13.11 | juned | Thank you very much |
| 13:13.46 | juned | how stupid I am :( |
| 13:14.33 | aiksa[LV] | these things happen |
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| 13:15.00 | juned | Yeah |
| 13:15.09 | aiksa[LV] | " No category context" basically tells you that asterisk configuration loader does not know what to do with this directive |
| 13:15.27 | aiksa[LV] | that "it just exists there in void" |
| 13:16.04 | juned | Ahh i see |
| 13:16.28 | juned | so here we can say our context is [mappings] i guess |
| 13:18.07 | aiksa[LV] | not context in dialplan sense |
| 13:18.27 | aiksa[LV] | every configuration file follows this logic |
| 13:18.50 | aiksa[LV] | you would most probably have [general] or [default] at the beginign of file |
| 13:19.10 | juned | Yeah it used to be there .. |
| 13:19.11 | aiksa[LV] | where you specify module wide configuration settings; |
| 13:19.25 | aiksa[LV] | think of sip.conf |
| 13:19.45 | aiksa[LV] | it also has that general part where you specify things like local network topology |
| 13:19.59 | aiksa[LV] | address to bind to, etc. etc. |
| 13:20.25 | aiksa[LV] | and after that there are seperate entries [XXXX] for each user and peer |
| 13:21.08 | juned | Yeah correct |
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| 13:21.13 | aiksa[LV] | but no configuration options exist completely outside configuration categroy |
| 13:21.59 | juned | I understood :) |
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| 13:30.52 | KKeXX | Hi |
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| 13:32.21 | KKeXX | I've a Macro what does's start/stop mixmonitor during a call. This macro is called via features.conf. This works fine, but to migrate the server to a newer version I've to rewrite that macro to gosub. Is it possible to call GoSub from features.conf? I've had no success with that .... |
| 13:32.35 | KKeXX | someone have experimented with that? |
| 13:33.24 | KKeXX | The macro call in features.conf: record => *2,self,Macro,record |
| 13:33.43 | KKeXX | The GoSub call: record => *2,self,Gosub(record,s,1) |
| 13:33.52 | KKeXX | But this does not work (for me) |
| 13:41.54 | newtonr | hmm |
| 13:44.12 | newtonr | Did you try record => *2,self,Gosub,"record,s,1" ? I'm curious if that works or not |
| 13:45.16 | newtonr | also what is the mode of failure? can you pastebin a debug log showing where it fails? |
| 13:45.30 | KKeXX | hmmm, think I haven't tried that one. |
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| 13:57.28 | KKeXX | @newtonr: There is not really a meaningfull entry in the log: http://pastebin.com/QfsbwBEP |
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| 14:11.57 | newtonr | KKeXX, I see "record,s,0" does a 0 priority exist ? |
| 14:12.48 | KKeXX | @newtonr, *hmmmm I've just seen it. No, it does not exist. But priority 1 is configured ... I'll check this |
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| 14:14.45 | newtonr | KKeXX, you might pastebin the whole dialplan as well |
| 14:15.02 | newtonr | or at least the relevant contexts |
| 14:19.50 | KKeXX | @newtonr, not that exciting: http://pastebin.com/XJemPQtq |
| 14:20.28 | KKeXX | This is just a Test |
| 14:22.38 | newtonr | weird, and if in features.conf you have record,s,1 I don't know why it would be looking for record,s,0 |
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| 14:23.01 | newtonr | KKeXX, if you can't figure it out or narrow it down further you might post a bug on issues.asterisk.org/jira with all your configs and debug log |
| 14:23.16 | newtonr | assuming you are using a recent, supported branch of asterisk |
| 14:24.02 | KKeXX | @newtonr, entry in features conf is record => *2,self,Gosub(apprecord,s,1) |
| 14:25.09 | KKeXX | current Version (where I tested) is 10.9 (not that recent), but planing to migrate to 11.19.0 |
| 14:25.17 | KKeXX | record => *2,self,Gosub(record,s,1) |
| 14:44.12 | newtonr | Yeah I would test in the latest 11 or 13 before posting a bug on tracker |
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| 17:06.06 | dennis_345 | Hello everybody. I have a problem with establishing connection between two asterisk servers using iax2. I am working on this problem for two days already. I run out of ideas and have no idea where to look further. It would be great if somebody could steer me in right direction. |
| 17:06.06 | dennis_345 | What do I want to achieve: I have two servers. One is server A with static IP address and DNS name iax.voip.domain.com . Other one is server B with dynamic IP behind NAT. I want to establish IAX2 connection between server A and B to be able to Dial from server A to B and visa versa. Here are iax.conf files for both servers: http://pastebin.com/2z9rUymi . Both servers are |
| 17:06.06 | dennis_345 | running Debian jessie and asterisk from Debian repositories (11.13.1~dfsg-2+b1). |
| 17:06.06 | dennis_345 | The problem is that registration does not happen. Status of peer is UNKNOWN when checked on server A and UNREACHABLE when checked on server B. Here are steps which I already did: I checked iax packets using tcpdump (tcpdump port 43762) on server A and I can see packets coming in and no packets going out. I checked that asterisk is listening for packets on UDP port 43762 (lsof |
| 17:06.06 | dennis_345 | -i -n | grep 43762) and indeed it does. I enabled iax debug in asterisk console using 'iax2 set debug on' and I do not see any packets or activity. That is as far as I got. I have no idea what could be a reason for such behaviour and how can I fix it. Could anybody help me? |
| 17:11.46 | [TK]D-Fender | username doesn't match |
| 17:12.04 | [TK]D-Fender | Also register => server_b:common_secret@iax.voip.domain.com:43762 <- why this number at the end? |
| 17:12.31 | dennis_345 | The number at the end is port number |
| 17:13.01 | [TK]D-Fender | Any reason not to use the standard? |
| 17:13.15 | [TK]D-Fender | Did you forward this port at each side? |
| 17:13.28 | [TK]D-Fender | But it remains the the name doesn't match |
| 17:13.35 | [TK]D-Fender | username=server_b |
| 17:13.42 | [TK]D-Fender | username=server_a |
| 17:14.20 | [TK]D-Fender | When you use a "username" the thing in [] doesn't matter.... |
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| 17:20.34 | dennis_345 | I didn't use standard port for security reasons and to avoid increasing server load because of brute-force attacks, I can change it to standard. What do you mean by port forwarding? Server A has public IP and doesn't have firewall. Server B is behind NAT and I cannot change NAT settings. I will try changing usernames to same values. |
| 17:21.46 | [TK]D-Fender | ok, username seems to be key there. Server A also didn't specify the port. |
| 17:21.47 | [TK]D-Fender | which you should anyway |
| 17:21.53 | [TK]D-Fender | if you're going to go that route |
| 17:22.05 | [TK]D-Fender | This might be OK. |
| 17:23.40 | [TK]D-Fender | the username should be a more "combined name" than mentioning the specific server. like "[trunk_between_a_and_b]" |
| 17:23.47 | [TK]D-Fender | logically. |
| 17:23.47 | [TK]D-Fender | not so long though. |
| 17:24.16 | [TK]D-Fender | because you'll ahve an entry on both servers, and on Server A it'd look funny to make an entry aying it is to connect TO Server A (itself) |
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| 17:27.40 | dennis_345 | I tried changing usernames and still no luck. Even with wrong usernames I would expect to see at least some feedback in console of server A, especially after 'iax2 set debug on' |
| 17:28.23 | [TK]D-Fender | check firewalls... |
| 17:28.53 | [TK]D-Fender | And set the port in your peers as I suggested |
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| 17:34.19 | dennis_345 | I switched to default port, still same result. How do I check firewalls? '# iptables -L -n -v -x' shows empty iptables for both servers. |
| 17:34.55 | [TK]D-Fender | iptables --list |
| 17:35.06 | [TK]D-Fender | always just go raw in case some syntax slips past you |
| 17:36.22 | dennis_345 | Still empty for both servers: http://pastebin.com/DzuVYS1a |
| 17:36.55 | [TK]D-Fender | ok, well lets look at teh IAX2 debug on the fixed host... |
| 17:41.18 | dennis_345 | That is the problem, I do not see any debug output on the fixed host even after 'iax2 set debug on'. And I can see arriving packets using tcpdump. Do you know how I can get any extra information from asterisk? |
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| 18:15.38 | cervajs2 | <PROTECTED> |
| 18:15.38 | cervajs2 | <PROTECTED> |
| 18:16.20 | Cuzner | dennis_345: core set verbose |
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| 18:42.13 | reveal | Unable to request channel DAHDI/0116 |
| 18:42.18 | reveal | what did i screw up with |
| 18:42.59 | WIMPy | Your argument to dial? |
| 18:43.13 | WIMPy | Is that really what you wanted to call? |
| 18:43.43 | reveal | its an ext on a system that is using the DID to call it |
| 18:43.52 | reveal | its a rj11 phone |
| 18:44.01 | reveal | with 0116 is mapped as a DID to ext 116 |
| 18:44.15 | WIMPy | That makes no sense. |
| 18:44.26 | reveal | its a pri |
| 18:44.51 | WIMPy | The 4th pri? |
| 18:45.14 | reveal | huh |
| 18:45.36 | reveal | we have a comdial pbx with a pri and its connected to asterisk |
| 18:45.39 | WIMPy | The 116th channel might be on the 4th pri. |
| 18:45.42 | reveal | via a card |
| 18:45.54 | reveal | its trying to dial it based on channels? |
| 18:46.05 | reveal | our pri only has 24 channels |
| 18:46.13 | WIMPy | That's what you seem to have done. |
| 18:46.30 | WIMPy | Well, in that case you'd need more than 4 PRIs :-) |
| 18:46.36 | reveal | right |
| 18:46.37 | reveal | we have 1 |
| 18:46.38 | reveal | LOL |
| 18:47.05 | reveal | dahdi show channels shows 47 psuedo chan ext |
| 18:47.39 | reveal | can i pick a channel like originate dahdi/5/0116 |
| 18:47.44 | WIMPy | You probably want to dial an extension on a group, not a channel. |
| 18:48.15 | WIMPy | You could do that. |
| 18:48.29 | reveal | im trying to get the asterisk pbx to dial my ext to put me in a meetme conf |
| 18:48.54 | WIMPy | But usually you don't want to select a cahnnel. Bad enough that DAHDI does so internally. |
| 18:49.05 | reveal | read the CLI output i notice it shows dahdi/i2/number and dahdi/i1/number answered |
| 18:49.32 | reveal | DAHDI/i2/0823-c4fa is proceeding passing it to DAHDI/i1/3072716 |
| 18:49.47 | WIMPy | You can't dial interfaces. You have to use groups. |
| 18:49.49 | reveal | i think thats from-pbx to ptsn right |
| 18:49.55 | *** join/#asterisk doome_ (~doome@78-131-72-25.pool.digikabel.hu) |
| 18:49.56 | reveal | oh |
| 18:50.11 | WIMPy | Possible, We don't know what is what. |
| 18:50.46 | reveal | DAHDI/i1/70866382-c4bf answered DAHDI/i2/2199377-c4fc |
| 18:50.47 | reveal | [2015-08-27 13:50:13] -- Native bridging DAHDI/i2/2199377-c4fc and DAHDI/i1/7086638226-c4bf |
| 18:50.58 | reveal | seems like its passing it off |
| 18:51.05 | reveal | either from the ptsn to the pbx or vice versa |
| 18:51.21 | reveal | ok so you also stated cant dial interfaces only groups |
| 18:51.47 | WIMPy | So you have two PRIs? One to telco and one to a PBX? |
| 18:52.04 | reveal | well |
| 18:52.10 | reveal | hold on me better explain |
| 18:52.13 | reveal | bare with me ok |
| 18:53.02 | reveal | we have a comdial dxp system with a pri card in it, that card is then connected to comcast ADTRAN and to asterisk, ADTRAN gives dialtone |
| 18:53.07 | reveal | does that make sense |
| 18:53.58 | reveal | using a t1 crossover between the pbx and asterisk |
| 18:54.58 | reveal | i have a A102 T1/PRI digi card in the asterisk machine |
| 18:55.47 | WIMPy | So that Adtran comes from your Telco? |
| 18:55.48 | *** join/#asterisk kolko (~kolko@46.48.58.17) |
| 18:55.51 | reveal | yes |
| 18:56.25 | reveal | the adtran uses B8ZS encoding and ESF for frame |
| 18:57.57 | reveal | our switch is NI-2 |
| 18:59.09 | reveal | WIMPy: did i do a decent enough job explaining it to you |
| 19:00.23 | WIMPy | Yes |
| 19:02.19 | reveal | ok good |
| 19:02.26 | reveal | sort of green as you can tell |
| 19:03.15 | WIMPy | You need to look at the examples dor DAHDI dial strings. You can find the groups there. |
| 19:03.46 | WIMPy | If your config was auto generated, you would have a group per interface already. |
| 19:04.03 | reveal | how do i check |
| 19:04.23 | reveal | i found it |
| 19:05.06 | reveal | http://pastey.org/view/e5f44073 |
| 19:05.49 | reveal | that? |
| 19:06.41 | reveal | here is another one |
| 19:06.42 | reveal | http://pastey.org/view/f2fc1b3b |
| 19:07.32 | WIMPy | Yes, there you have groups 0 and 1 for the two interfaces. |
| 19:08.08 | reveal | what does that mean, is that bad retarded ok or... |
| 19:08.22 | WIMPy | And a rather senseless Group 11 for both. |
| 19:08.33 | reveal | you mean the 0,11 and 1,11 |
| 19:08.35 | reveal | LOL |
| 19:08.56 | WIMPy | That means you can dial group 0 or 1 depending on which interface you want to send a call to. |
| 19:10.10 | reveal | well |
| 19:10.28 | reveal | the PBX is plugged into port B on the card in asterisk and port A goes to the adtran |
| 19:11.04 | reveal | i assume from-pbx is from our pbx and from-ptsn is from the adtran that provides dialtone |
| 19:11.15 | reveal | port 1 is group 0 and port 2 is group 1 |
| 19:11.26 | reveal | group 0 would dial DIDs right? |
| 19:12.01 | WIMPy | yes |
| 19:12.27 | WIMPy | Group 0 would call the PSTN and group 1 would call extensions on your PBX. |
| 19:12.43 | reveal | so if i wanted the asterisk to spawn a call on the PBX ext 0116 |
| 19:12.47 | reveal | how would that work |
| 19:14.34 | reveal | exten => _0XXX,1,dial(dahdi/g1/${EXTEN}) |
| 19:14.35 | reveal | ? |
| 19:14.56 | WIMPy | Something like that. |
| 19:15.08 | reveal | LOL |
| 19:15.20 | WIMPy | If your internal extensions start with 0. |
| 19:15.36 | reveal | on our hard phones no i mapped it as a DID to 0116 |
| 19:16.00 | reveal | can i do dialplan show 0116@default |
| 19:16.35 | reveal | '_0XXX' => 1. dial(DAHDI/G1/${EXTEN}) [pbx_config] |
| 19:16.39 | reveal | thats what that says |
| 19:16.48 | reveal | for dialpan show 0116@default |
| 19:20.01 | reveal | can i do originate dahdi/g1/0116 |
| 19:20.27 | WIMPy | yes |
| 19:20.54 | reveal | that will dial 0116 from asterisk right |
| 19:21.08 | WIMPy | yes |
| 19:21.20 | reveal | then to hangup i type hangup? |
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| 19:22.36 | reveal | says two ways to do it |
| 19:22.54 | reveal | when i did originate dahdi/g1/100 |
| 19:25.14 | reveal | ok it dialed me says sorry thats not a valid extension please try again |
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| 20:50.01 | [NC] | We have a client with Asterisk 11.17.1 (FreePBX 2.11) where T.38 sends a single no-signal udptl message to the remote provider even though the ATA itself sends many such message. How can we get Asterisk to send multiple no-signal too (or forward all of those it gets from the ATA)? |
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