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| 03:02.06 | shaggycat | Hi all! I have problem with permits incoming calls via _SIP_ |
| 03:02.43 | shaggycat | If I setting up caller-id using sip.conf, caller-id of incoming calls, it works |
| 03:03.01 | shaggycat | But without it call to special extension does not works |
| 03:04.47 | shaggycat | And I see in console: Call from '' to extension '950' rejected because extension not found. |
| 03:07.04 | shaggycat | http://pastebin.com/ZtjQSgC1 - it does not work |
| 03:07.07 | shaggycat | dialplan |
| 03:09.06 | shaggycat | Is it possible to use http://pastebin.com/WC8xXqVN in sip.conf this syntaxis? |
| 03:16.42 | shido6 | no |
| 03:16.56 | shido6 | you need a valid caller*id there |
| 03:17.06 | shido6 | brb |
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| 03:32.12 | shaggycat | shido6: but how can I doo? |
| 03:32.15 | shaggycat | *do |
| 03:32.24 | shido6 | what is the ask? |
| 03:32.29 | shido6 | I missed it. |
| 03:32.47 | shaggycat | I need to get incoming calls with different caller-ids |
| 03:33.12 | shaggycat | Provider's server does not support registration in server |
| 03:33.36 | shaggycat | And if number has incoming call, provider's server call to my server |
| 03:34.34 | shido6 | do you want to direct inbound calls with different callerid's to different places? |
| 03:35.52 | shido6 | please rephrase what you are looking to do |
| 03:36.50 | shaggycat | I want to get direct inbound calls with different caller id's, routed in my special context - for incoming calls (queues, etc) |
| 03:37.01 | shido6 | ok. |
| 03:37.10 | shido6 | inbound via SIP ? |
| 03:37.14 | shaggycat | yes |
| 03:37.37 | shido6 | multiple providers ? single provider? |
| 03:37.47 | shaggycat | single provider |
| 03:38.25 | shido6 | ok - so you'll need to create the contexts you want then using a number pattern match, direct those calls to their respective contexts |
| 03:38.37 | shaggycat | Call from '' to extension 'myext' rejected because extension not found. |
| 03:38.41 | shaggycat | I get in logs |
| 03:38.46 | wyoung | hey, I am having issues with ticking and jittering noises occuring hwne in a SIP call |
| 03:38.52 | shaggycat | But this context exist |
| 03:38.56 | shido6 | good - so you know what its looking for |
| 03:39.06 | wyoung | it doesn't happen when phoning other extensions in the same network and it doesn't happen with some calls to mobile phones |
| 03:39.11 | wyoung | how does one troubleshoot this? |
| 03:39.12 | shido6 | but I need to know more about 'myext' |
| 03:39.23 | wyoung | [TK]D-Fender: you in ? |
| 03:39.38 | shido6 | show us the call ( use pastebin.com ) |
| 03:40.18 | shido6 | shaggycat - show the call in pastebin.com |
| 03:40.51 | shaggycat | shido6: this the part of extension file |
| 03:41.10 | shido6 | please do not show the contents here just the URL |
| 03:41.22 | shaggycat | http://pastebin.com/Pv0ji0ap |
| 03:42.01 | shido6 | wow - you weren't kidding you are actually using 'myext' |
| 03:42.31 | shido6 | myext doesnt represent a number, does it? is this exactly whats in the file? |
| 03:42.44 | shido6 | did you change anything? [ did you mess with the evidence? ] |
| 03:43.29 | shaggycat | http://pastebin.com/CjXTvKvz and it is a part of sip.conf |
| 03:43.45 | shaggycat | shido6: I also tryed to use number - 950 |
| 03:43.54 | shaggycat | and it doesn't work same |
| 03:44.18 | shido6 | and how are you calling it? |
| 03:44.22 | shido6 | show the sip debug |
| 03:44.26 | shaggycat | I use puppet tool for autogenerate zabbix configs (I'm autor of code), |
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| 03:45.34 | shido6 | where are the other patterns you are attempting to match? |
| 03:45.52 | shido6 | http://pastebin.com/Pv0ji0ap only shows 'myext' where are the others? |
| 03:46.03 | shaggycat | log is too long |
| 03:46.15 | shaggycat | I need a time to paste it in pastebin |
| 03:46.47 | shido6 | ok so you have something like myext2 or otherext in this twilio_moncton_xeniya context? - perhaps they exist in internals or redirects? |
| 03:48.30 | shido6 | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching look at the patterns and how they are being directed to different places - this should help |
| 03:48.32 | shaggycat | http://pastebin.com/s3BUXAay |
| 03:48.38 | shaggycat | It's debug log |
| 03:49.34 | shaggycat | 10.0.5.225 and 10.0.5.226 it's internal IPs of two asterisk servers (i use registration in closest server to outgoing calls) |
| 03:49.45 | shido6 | i see that - and thats not going to work unless you dialed myext@10.0.5.225 |
| 03:50.04 | shido6 | see line 26? 404 not found? |
| 03:51.15 | shido6 | ok i see line 155 |
| 03:51.22 | shido6 | 159 |
| 03:51.39 | shaggycat | yes, I seen |
| 03:52.04 | shaggycat | but I know.. If I add specific extension, |
| 03:52.12 | shaggycat | +266696687 |
| 03:52.15 | shaggycat | It works |
| 03:52.52 | shaggycat | and specific phone in sip.conf |
| 03:52.53 | shido6 | you want only alpha characters and not numeric extensions? |
| 03:53.08 | shaggycat | shido6: I used numeric too |
| 03:53.13 | shaggycat | same problem |
| 03:53.20 | shaggycat | It's dialplan problem |
| 03:53.22 | shaggycat | No sip |
| 03:53.41 | shaggycat | And I don't undersrand why my asterisk don't route my calls |
| 03:53.48 | shaggycat | *incoming calls |
| 03:54.21 | shido6 | well I generally use "()" around my arguments |
| 03:55.07 | shido6 | what does the cli debug tell you when you call in? |
| 03:55.16 | shido6 | exten => myext,1,Wait(1) |
| 03:55.31 | shaggycat | http://pastebin.com/9BT7P8tn |
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| 03:55.35 | shaggycat | It's dialplan |
| 03:56.09 | shaggycat | if I call outside the server? |
| 03:57.01 | shaggycat | asterisk01*CLI> Call from '' to extension 'myext' rejected because extension not found. |
| 03:57.08 | shaggycat | when I used digest ext, |
| 03:57.26 | shaggycat | It was asterisk01*CLI> Call from '' to extension '950' rejected because extension not found. |
| 03:58.08 | shaggycat | look, can I show you working configs? It works only with one caller-id |
| 03:58.21 | shaggycat | But I need to have works it with different |
| 03:58.45 | shido6 | sure |
| 03:58.51 | shido6 | please do |
| 03:58.59 | shaggycat | I need to generate it by puppet |
| 03:59.05 | shaggycat | Just some of minutes |
| 03:59.12 | shido6 | ok |
| 03:59.32 | shido6 | I need to make some coffee - it will take 7 minutes. |
| 03:59.50 | joj[] | Hi. Porting from 1.8 to 13, I notice that in 13, cdr_pgsql fails to insert CDR records for dispostion = 'NO ANSWER' because for the "answer" SQL field, it passes an empty string and not NULL, making postgres fail because of a cast error. setting default values for that field doesn't work because the value passed is not NULL. I don't see any config option to change this in cdr_pgsql. Is creating a trigger |
| 03:59.53 | joj[] | and rewrite the value before insert the best solution, or are other people handling this differently? I didn't find any documentation on this either. |
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| 04:07.29 | shaggycat | shido6: http://pastebin.com/TK6pPSpV it works only with incoming calls Skype :( |
| 04:08.02 | shaggycat | +266696687 is the skype-out caller id |
| 04:08.31 | shaggycat | shido6: thank you very much for helping |
| 04:08.45 | shido6 | ah its in internals |
| 04:08.58 | shido6 | and also in twilio_moncton_xeniya |
| 04:09.17 | shido6 | so it hits internals first |
| 04:10.54 | shaggycat | shido6: I include internals to have queues |
| 04:11.04 | shaggycat | *to have working queues |
| 04:11.49 | shaggycat | and I'm fear to broke security |
| 04:12.54 | shaggycat | shido6: could you please tell, What is the corect solution? |
| 04:13.18 | shido6 | im still trying to wrap my brain around why you have 150 and 151 there |
| 04:14.13 | shaggycat | shido6: it's something generated by puppet |
| 04:14.21 | shaggycat | May be it doesn't required |
| 04:14.34 | shaggycat | And some not required logic |
| 04:15.07 | shaggycat | I can try to delete 136 and 137 |
| 04:15.14 | shaggycat | and check how skype works |
| 04:15.17 | shido6 | dont delete anything - show me the call |
| 04:17.42 | shaggycat | shido6: http://pastebin.com/ka0mxkL3 |
| 04:20.26 | shido6 | ok so where's the other queue? |
| 04:20.53 | shido6 | i see company_queue |
| 04:21.48 | shido6 | didnt you want to send another call type that uses a different pattern/extension sent to a different place? |
| 04:22.14 | shaggycat | shido6: in this config queue is one. |
| 04:22.20 | shaggycat | It's generated by puppet |
| 04:22.31 | shaggycat | Why do you think it's important? |
| 04:22.51 | shaggycat | This call works, I hang up after queue started |
| 04:23.11 | shaggycat | But if I don't setting up in extensions.conf and in sip.conf specific caller-id, |
| 04:23.16 | shaggycat | calls doesn't works |
| 04:23.57 | shido6 | so you want to change line 406 callerid="+266696687" in http://pastebin.com/TK6pPSpV ? |
| 04:24.02 | shaggycat | +266696687 - this caller id |
| 04:24.30 | shaggycat | shido6: I want to recive calls from different caller ids |
| 04:24.40 | shaggycat | Not only test caller-id +266696687 (skype) |
| 04:25.08 | shaggycat | but it works only if I add specific extension :( +266696687 |
| 04:25.09 | shido6 | show me the call |
| 04:25.20 | shido6 | ok i see it - stdby |
| 04:25.24 | shaggycat | which call? |
| 04:26.27 | shaggycat | so it works if I delete lines: |
| 04:27.14 | shaggycat | 136 and 137 in http://pastebin.com/TK6pPSpV |
| 04:27.45 | shaggycat | exten => +266XX.,1, Macro(stdexten,+266696687,SIP/+266696687) |
| 04:27.45 | shaggycat | exten => +266XX.,1, Goto(+266696687|1) |
| 04:27.55 | shaggycat | with this lines |
| 04:28.07 | shido6 | ok so what numbers are you expecting ? |
| 04:28.37 | shido6 | match those expected patterns and instead of using the number add ${EXTEN} |
| 04:29.00 | shaggycat | +1.... |
| 04:29.04 | shaggycat | +1XXXX. |
| 04:29.12 | shido6 | ok |
| 04:29.30 | shaggycat | but use it in one pattern with skype, |
| 04:29.36 | shaggycat | may be better for tests |
| 04:29.43 | shido6 | I'll add my bits , stdby |
| 04:29.49 | shaggycat | +[12]XXX. |
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| 04:32.05 | shaggycat | If I edit in sip.conf name of peer to [+26XX.] |
| 04:32.10 | shaggycat | It doesn't works |
| 04:32.19 | shido6 | no thats not going to work |
| 04:33.06 | shido6 | i would try with line 129 and change +1XXXX. to ${EXTEN} in the last 2 instances of +1XXXX. |
| 04:33.25 | shido6 | that stdexten tho is going to make things weird |
| 04:33.54 | shido6 | exten => +1XXXX.,1, Macro(stdexten,+1XXXX.,SIP/+1XXXX.) exten => +1XXXX.,1, Macro(stdexten,${EXTEN},SIP/${EXTEN}) |
| 04:34.25 | shaggycat | But I haven't any test trunc :( |
| 04:34.34 | shaggycat | With +1 caller id |
| 04:34.50 | shido6 | exten => +1XXXX.,1, Macro(stdexten,+1XXXX.,SIP/+1XXXX.) exten => +1XXXX.,1, Macro(stdexten,${EXTEN},SIP/${EXTEN}@remotegatewayhere) |
| 04:35.36 | shido6 | what device are you trying to call? |
| 04:35.42 | shaggycat | skype |
| 04:35.48 | shaggycat | and skype-out |
| 04:36.02 | shaggycat | I think better to use pattern with skype caller-id |
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| 08:40.56 | shaggycat | can anybody with North America's phone number dial my asterisk server? |
| 08:41.27 | shaggycat | Just to numbers in North America (+1XXX.) area |
| 08:41.32 | shaggycat | *just 2 |
| 08:43.23 | zappaFan | Hi there everybody, I was wondering if there's a website in which I could test SIP trunk configuration, make and receive international calls for free. |
| 08:43.47 | zappaFan | I mean, just for testing purposes, for a couple of days.. |
| 08:46.29 | Alblasco1702 | zappaFan, maybe you can use voipbuster for it. |
| 08:51.29 | zappaFan | Alblasco1702: thanks a lot!, I just register, will see how it works |
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| 08:51.36 | zappaFan | *registered |
| 08:51.48 | Alblasco1702 | zappaFan, YW |
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| 10:34.09 | jkroon | is it possible to select multiple values using func_odbc in a single statement? |
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| 10:58.28 | jkroon | http://www.voip-info.org/wiki/view/Asterisk+func+hash |
| 10:58.30 | jkroon | thanks. |
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| 11:02.37 | WIMPy | Never noticed that one. |
| 11:06.12 | pa | GV down again |
| 11:06.13 | pa | asdasd |
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| 11:39.09 | cervajs2 | is there someone for consultation of side effects of https://github.com/asterisk/asterisk/commit/83dadc4683abcd10f0c4566abef541d997dcf5b8 ? |
| 11:43.44 | file | what's the question? |
| 11:45.49 | cervajs2 | i have applied this patch to 13.9.1 and i see some random problems with "[Jun 23 13:31:27] DEBUG[12419][C-00000049]: app_queue.c:4232 can_ring_entry: Local/300@route_phones_1 has another call trying, can't receive call" |
| 11:45.59 | cervajs2 | but the agent is free. hints are ok |
| 11:46.31 | file | you'd need to get a log preceeding and see |
| 11:47.38 | zappaFan | completely off topic but was wondering if there's a channel where I could ask questions about laptop hardware upgrading/tweaking? |
| 11:47.48 | cervajs2 | it's problem to replicate. my first question is if someone else is seeing this. nothing similar in issue tracker was found |
| 11:48.05 | file | haven't seen anything |
| 11:48.39 | zappaFan | I haven't been to irc channels since the early 2000's, glad it's still going on, this channel seems quite useful |
| 11:48.50 | cervajs2 | file: ok. will try found scenario for replication |
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| 11:50.08 | zamba | zappaFan: #hardware |
| 11:50.57 | zappaFan | nice, thanks a lot! |
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| 13:47.40 | Guest1066 | Can anyone share some documentation of how to use a master/slave database configuration with asterisk cdr? Currently using odbc with asterisk. I fail to find this documentation some how. |
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| 13:48.24 | Guest1066 | I've added all the databases to odbc.ini but now I am up to configure asterisk for it. |
| 13:48.35 | Guest1066 | I have one database set up today |
| 13:49.22 | Guest1066 | Working with asterisk sort to say. |
| 13:51.10 | *** join/#asterisk obelixBE (~obelix@109.142.230.133) |
| 13:51.25 | Samot | The slave doesn't talk to Asterisk. |
| 13:51.29 | Samot | It talks to the master. |
| 13:51.35 | Guest1066 | http://codepad.org/VxMvSMsY |
| 13:51.38 | Guest1066 | My configurations |
| 13:51.46 | Guest1066 | Yes Samot, but if the master go down asterisk must know this |
| 13:51.51 | Guest1066 | and push to the slave |
| 13:52.21 | Samot | It will only push to where you tell it. |
| 13:52.31 | Samot | It doesn't know if A is down go to B. |
| 13:53.00 | Guest1066 | So it has no db failover functionality at all? |
| 13:53.37 | Guest1066 | Like SIPPEER? :) |
| 13:53.43 | [TK]D-Fender | Guest1066, You defined 3 connections in odbc.ini .... and then didnt' bother creating multiple CDR tnreies to USE them, |
| 13:53.56 | [TK]D-Fender | Thre is no failover. |
| 13:54.00 | [TK]D-Fender | there is only MULTIPLE |
| 13:54.10 | Guest1066 | Meaning it will write to all of them? |
| 13:54.13 | [TK]D-Fender | yes |
| 13:54.13 | Guest1066 | If I specify three |
| 13:54.32 | Guest1066 | If one go down it will not cause any issues? Just continue to mawl data to the other one? |
| 13:54.41 | [TK]D-Fender | it goes to all |
| 13:54.45 | [TK]D-Fender | if one fails, then one fails |
| 13:55.02 | [TK]D-Fender | Why would hte LOCALHOST one fail? |
| 13:55.08 | Guest1066 | That is the current one |
| 13:55.12 | Guest1066 | it will be removed |
| 13:55.20 | [TK]D-Fender | Why? |
| 13:56.07 | Guest1066 | Well maybe you are right, no harm in keeping it as a last resort. |
| 13:56.49 | [TK]D-Fender | wrong approach completely |
| 13:57.17 | [TK]D-Fender | It should be the ONLY place you put these and you should set up replication between DB's to handle backup. |
| 13:57.34 | [TK]D-Fender | So that * doesn't get lagged trying to write to everything itself |
| 13:57.34 | Samot | You need to define your other three connections in res_odbc.conf |
| 13:57.38 | Samot | Or other two. |
| 13:58.31 | Guest1066 | I asked earlier Fender if one go down will it cause any issues, such as lagg. |
| 13:58.36 | Guest1066 | We dont want that |
| 13:58.49 | [TK]D-Fender | keep everything local and let the DB do it's job |
| 13:58.58 | Guest1066 | Sounds sound |
| 14:06.53 | ||cw | the kinds of things that would make the local DB go down will very likely take out asterisks too. IMO you're worrying about the wrong thing |
| 14:07.38 | ||cw | use raid, ECC ram, UPS power backup, standby replication, etc. |
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| 14:09.54 | zappaFan | here's how elastix implements RAID, idk, might help you: http://www.elastix.com/elastix-raid-setup-step-by-step-including-recovery/ |
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| 14:11.15 | igcewieling | Guest1066: Asterisk will lag badly if it cannot write to the database (CDR, CEL, etc) |
| 14:15.30 | Guest1066 | Cool |
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| 14:18.52 | ||cw | zappaFan: that's just your standard linux software raid |
| 14:18.55 | Guest1066 | We do not want that |
| 14:19.53 | ||cw | mdraid mirroring is quite fast and robust. for anything above mirroring, hardware is better |
| 14:20.34 | ||cw | a raid card with cache and BBU will get you an IO performance gain as well, but I'm not sure that's important for this workload |
| 14:21.30 | [TK]D-Fender | mdraid = write hole |
| 14:21.40 | [TK]D-Fender | if you're going SW then it's ZFS or bust |
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| 14:22.27 | ||cw | now that it's stable, yeah |
| 14:25.59 | igcewieling | You will find that all your redundancies and fail-overs didn't work when you needed them. The best you can hope for is to take reasonable steps such as RAID redundant power supplies. |
| 14:27.26 | ||cw | and backups. and test your restore procedures |
| 14:28.06 | ||cw | hot standby systems are nice and all, but Ive found it's often faster to fix the primary than to actually do the switchover |
| 14:28.26 | ||cw | unless you go for fancy expensive clustering stuff |
| 14:31.46 | igcewieling | I find it funny when customers want a "standby" server. They *never* work right. |
| 14:32.30 | igcewieling | "What do you mean syncing from primary to standby broke 6 months ago and nobody noticed until our main server burst into flames?" |
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| 14:42.20 | igcewieling | The iptables rules I use to keep out sip scanners, in case they are useful to someone: http://pastebin.com/xXYkxqR7 |
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| 14:44.40 | WIMPy | That's more that I've ever seen. |
| 14:45.27 | flujan | Hello. is it possible to use Dial do dial to a sipuri instead of a extension? I am trying to use netborder call analyser. Need to dial something like this sip:4444@IP;cpd=on |
| 14:46.12 | WIMPy | You never Dial() an extension unless you're using a local channel. |
| 14:49.11 | igcewieling | flujan: Dial(SIP/telephonenumber@ipaddressorhost) |
| 14:49.47 | flujan | igcewieling: thanks for the tip. |
| 14:49.56 | igcewieling | that dials by ip address, but I doubt it will do what you want. |
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| 15:25.26 | x-fak | hi |
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| 16:52.14 | trelane | how do I get debug output that normally goes to the asterisk cli/console in AMI? |
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| 17:07.01 | [TK]D-Fender | You don't |
| 17:10.04 | newtonr | trelane, yeah, you can't |
| 17:10.23 | newtonr | trelane, you CAN get the cli/console output in a log file. See logger.conf |
| 17:10.37 | newtonr | https://wiki.asterisk.org/wiki/display/AST/Logging |
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| 20:28.14 | epaphus | hello all. Could anybody please provide me an example of what line i would have to add to my extensions.conf so that calls from ext 100 when ringing ext 101 will ring with a special ringtone? |
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| 20:30.11 | WIMPy | Where does your 101 go? |
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| 20:30.47 | WIMPy | What kind of phone(s?) does it ring? |
| 20:31.30 | epaphus | actually, i think this is easier. I already have a queue() |
| 20:31.36 | epaphus | so i just need to apply the ringtone to that queue |
| 20:31.44 | epaphus | as default for all the members |
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| 21:19.34 | csshih | anyone else having issues with google voice and openssl? |
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| 22:07.14 | drmessano | csshih: is it down again? |
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| 22:28.22 | csshih | drmessano, yes |
| 22:28.46 | csshih | PIAF forum reports the same thing: http://pbxinaflash.com/community/threads/gv-stopped-connecting-today.19394/page-5 |
| 22:35.07 | drmessano | I think the end is nigh |
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| 22:35.54 | crised | does anyone knows about a voip service that has servers in Chile, South America? |
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| 23:14.38 | csshih | drmessano, again? :D |
| 23:19.34 | drmessano | I dunno |
| 23:19.44 | drmessano | It's any time now, really |
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| 23:50.39 | wyoung | hey, so any ideas? |
| 23:50.42 | wyoung | [TK]D-Fender: ? |