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| 01:15.17 | lvlinux | does mixmonitor have to be put AFTER answer() in the dialplan, or will it just start from the answer() when it hits it? |
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| 01:36.09 | [TK]D-Fender | "core show application mixmonitor" |
| 01:37.26 | MaliutaLap | [TK]D-Fender: we should get matching tactical turtle necks ;) |
| 01:37.37 | lvlinux | [TK]D-Fender: yes of course, sorry i forgot to check there lol |
| 02:01.33 | lvlinux | Hmmmm, how can I have * use directmedia only when the call doesn't need to be recorded? |
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| 02:03.53 | lvlinux | do I have to use Answer() before Dial()?? |
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| 02:06.00 | [TK]D-Fender | It will TRY to use it if it's allowed and not prevented |
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| 02:09.07 | lvlinux | yes i understand that---is there a way to disable directmedia on a per-call basis? |
| 02:10.04 | lvlinux | without using Answer()? |
| 02:10.58 | [TK]D-Fender | <[TK]D-Fender> It will TRY to use it if it's allowed and not prevented <- |
| 02:11.13 | lvlinux | The reason I'm trying to avoid answering the call before dialing is to keep the caller from hearing the ringing studder when * picks up the call. |
| 02:12.06 | lvlinux | [TK]D-Fender: so is there a way to prevent it in the dialplan, other than using Answer()? |
| 02:12.51 | [TK]D-Fender | You really need to get your head clear about this... |
| 02:13.01 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> It will TRY to use it if it's allowed and not prevented <- |
| 02:14.28 | lvlinux | Yes I understand that (I think lol). So can i PREVENT it via the dialplan? |
| 02:16.35 | lvlinux | I know that right now it is trying directmedia, since I have a dial() with nothing before it that stops directmedia. When the callee answers, * trys and sees that it doesn't need to be in the path, so it uses directmedia. |
| 02:17.58 | lvlinux | Am I confused about this? |
| 02:18.03 | lvlinux | lol |
| 02:18.54 | lvlinux | What I'm wondering, is if there is some sort of variable I can set in the dial string or somewhere on this extension that will prevent * from trying to use directmedia. |
| 02:19.50 | lvlinux | on this particular extension, not all the time. |
| 02:32.53 | [TK]D-Fender | Anytime * has a REASON to be in the middle it will force it |
| 02:33.03 | [TK]D-Fender | Give it a reason |
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| 02:58.17 | lvlinux | and is there a reason besides Answer()?? |
| 03:05.22 | [TK]D-Fender | If * has to LISTEN for anything |
| 03:05.24 | [TK]D-Fender | or RECORD |
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| 03:52.13 | lvlinux | [TK]D-Fender: well I put Mixmonitor() before the dial and it worked. The reason I didn't think to do that before is that according to "core show application mixmonitor" you have to have an answer first. So I thought it wouldn't work putting it before the dial string (and thus the answer of the call). So it's like I originally asked earlier this evening, mixmonitor starts recording whenever the call is answered, regardless of whether it is before the d |
| 04:06.12 | [TK]D-Fender | It records as soon as there is any audio to record |
| 04:06.29 | [TK]D-Fender | if you answer, THEN dial ... then obviously there is time at the front end |
| 04:06.35 | [TK]D-Fender | This is simple math |
| 04:06.41 | [TK]D-Fender | and it's all in the sintructions. |
| 04:07.17 | [TK]D-Fender | Which if you don't want it recording ... unless there is something WORTH recording.. then you should read mixmonitor's instru tions |
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| 12:23.38 | fordfrog | hi, i have a lot of warnings on my asterisk (11.17.1) related to nat i guess. there is a private network with firewall with sip support. ip of the firewall is 10.14.101.3. there is another firewall on the path where public ip address is defined (i do not have control of that one). asterisk sees all inner peers with ip address 10.14.101.3. these are the messages i get in the log: [Jan 5 13:20:13] WARNING[27661]: chan_sip.c:3755 |
| 12:23.38 | fordfrog | __sip_xmit: sip_xmit of 0x7f9658040f20 (len 545) to 10.14.101.3:1024 returned -1: Operation not permitted |
| 12:23.47 | fordfrog | any idea how to fix the issue? |
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| 12:24.50 | WIMPy | You don't seem to have a route. |
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| 12:26.30 | fordfrog | WIMPy, yes, there is no route from the asterisk to the private ip address as the server is in internet and the clients are in a private network |
| 12:27.02 | fordfrog | i guess what i need is to make asterisk see the public ip address that is on the provider gateway instead of the private gateway |
| 12:28.02 | WIMPy | Or just make it ignore the IP. There are tons of guide on how to configure Asterisk for NAT situations. see nat= in your sip.conf. |
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| 12:28.49 | fordfrog | i found guide that still proposes nat=yes which is obsolete for long time i guess. will try to look more |
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| 12:46.07 | WIMPy | There is nothing obnsolete about NAT. |
| 12:46.16 | WIMPy | No more than SIP as a whole :-) |
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| 13:54.43 | fordfrog | WIMPy, i meant nat=yes. for some unknown reason it did not work for me when i changed it to nat=force_rport,comedia but now it does |
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| 14:24.03 | iresf | hi everyone |
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| 14:25.27 | iresf | i have installed asterisk on debian and also zopier but i donw know what ip asterisk is ? |
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| 14:27.47 | iresf | anyone here to help me ? |
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| 14:47.41 | [TK]D-Fender | What is your actual question? |
| 14:50.11 | iresf | [TK]D-Fender : i configured sip.conf then i assigned host = dynamic but in zoiper i can not create a sip account |
| 14:50.35 | [TK]D-Fender | ....because? |
| 14:51.49 | iresf | [TK]D-Fender : why ? |
| 14:51.59 | [TK]D-Fender | I'm asking YOU. |
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| 14:52.08 | [TK]D-Fender | You said you can't create an account in ZOIPER |
| 14:52.13 | [TK]D-Fender | Watch what you are saying |
| 14:52.20 | [TK]D-Fender | What ERROR does it give you? |
| 14:52.33 | [TK]D-Fender | What is STOPPING you from creating it? |
| 14:53.36 | iresf | [TK]D-Fender : i entered in domain an this ip : 192.168.1.10 and my ip host is 192.168.0.100 |
| 14:55.06 | [TK]D-Fender | And are those IP's RIGHT? |
| 14:55.19 | [TK]D-Fender | is that 1 subnet? 2? |
| 14:55.46 | iresf | [TK]D-Fender : no |
| 14:55.51 | [TK]D-Fender | 3? |
| 14:55.58 | [TK]D-Fender | 12? |
| 14:57.51 | [TK]D-Fender | How does this relate to not being able to create an account entry in Zoiper? |
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| 15:02.34 | iresf | [TK]D-Fender : zoiper installed on my host and asterisk also installed on same host is it right ? |
| 15:03.39 | [TK]D-Fender | Why would there be TWO IP's there? |
| 15:04.49 | iresf | because i read asterisk quick start guide |
| 15:05.13 | [TK]D-Fender | how does reading that guide suddenly give you TWO different IP's on your server? |
| 15:05.34 | [TK]D-Fender | We are down to having a basic grasp of networking. |
| 15:06.15 | iresf | asterisk server ip is host ip ? |
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| 15:08.28 | iresf | how to find asterisk server ip ? |
| 15:08.52 | iresf | i dont know what is asterisk server ip |
| 15:08.56 | mrfancypants | It's the address of the host that you have asterisk installed on. Run IFCONFIG. |
| 15:10.05 | iresf | mrfancypants : yes it is 192.168.0.100 but when i entered it zoiper it does not create sip account |
| 15:13.11 | iresf | mrfancypants : when i enter 192.168.0.100 as domain in zoiper to create sip account , zoiper can not create sip account |
| 15:14.50 | [TK]D-Fender | Make sure Zoiper is using a DIFFERENT port from the standard |
| 15:15.05 | [TK]D-Fender | Because otherwise they will both fight for the same standard port |
| 15:15.13 | WIMPy | iresf: You cannot run multiple SIP software on the same host without modifying port numbers. |
| 15:15.49 | iresf | WIMPy : yes that's right |
| 15:16.01 | WIMPy | iresf: But if you already have trouble finding your IP, using Asterisk will give you some very frustrarting weeks. |
| 15:16.32 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:17.32 | WIMPy | Using VoIP requires some basic understanding of IP. |
| 15:18.09 | WIMPy | And VoIp/telephony/PBXs/software are not the easy ones to configure. |
| 15:18.35 | WIMPy | ~book |
| 15:18.35 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 15:22.08 | iresf | WIMPy : is it not related to asterisk -r ? |
| 15:23.13 | WIMPy | What is related to that? |
| 15:23.33 | WIMPy | Your questions seem rather random. |
| 15:24.28 | WIMPy | We can't look in to your head, so make sure to ask questions that make sense to others. |
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| 15:31.33 | aurs | :D |
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| 15:35.58 | PoWeRKiLL | hi |
| 15:36.22 | PoWeRKiLL | how can I get the real exten when a call come into fax, hint extension ? |
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| 15:38.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:41.06 | [TK]D-Fender | That makes no sense. Please rephrase your question |
| 15:41.33 | [TK]D-Fender | Nothing "calls" a hint. |
| 15:41.45 | [TK]D-Fender | "hint" is a prioirty |
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| 15:46.07 | stkoch | hi back... |
| 15:47.44 | stkoch | I am at porting the chan_lantiq from asterisk 1.8 to asterisk 11 (and then 13) |
| 15:48.26 | stkoch | This is chan_lantiq for asterisk 1.8: http://paste.debian.net/360127/ |
| 15:48.55 | stkoch | The problem are the AST_FORMAT_ULAW, ... specifiers |
| 15:48.59 | [TK]D-Fender | You should be keeping this in #asterisk-dev |
| 15:49.13 | [TK]D-Fender | Youare doing devel work, not standard usage |
| 15:49.18 | [TK]D-Fender | There is a channel for that |
| 15:49.38 | stkoch | Ok, I write to it |
| 15:49.46 | stkoch | For completness: |
| 15:49.59 | stkoch | There is a table rtpPTConf.nPTup[IFX_TAPI_COD_TYPE_MLAW] = AST_FORMAT_ULAW; |
| 15:50.19 | stkoch | but these does not allow values greater than int8_t |
| 15:50.45 | stkoch | So that it work it is needed to change #define AST_FORMAT_INC 100000 to #define AST_FORMAT_INC 100 |
| 15:50.58 | [TK]D-Fender | This isn't the place |
| 15:50.59 | stkoch | but thats not the solution... |
| 15:51.02 | [TK]D-Fender | take it to -dev |
| 15:51.06 | stkoch | OK |
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| 15:56.13 | PoWeRKiLL | [TK]D-Fender when I got a fax on exten => fax,1 how do I get the real dialed number ${EXTEN} is equal to fax and I want to get the real dialed number |
| 15:57.06 | [TK]D-Fender | Either push that onto another variable at the start of your call or look at CDR() |
| 15:59.06 | PoWeRKiLL | you mean it's starting on exten => X.,1, then go to fax ? if yes I can Set(REALNUM=${EXTEN}) |
| 16:01.56 | [TK]D-Fender | "you" don't go to fax. * jumps automatically if you told it to detect |
| 16:04.27 | [TK]D-Fender | And anytime before then... you can do whatever you want... including setting other variables |
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| 16:26.25 | Kobaz | hmm |
| 16:26.35 | Kobaz | anyone know if you can disable hanging up of calls on a polycom |
| 16:28.09 | WIMPy | That is not a feature I've ever seen. |
| 16:28.45 | cmendes0101 | just curious what would be the reason? |
| 16:29.37 | ipn | does anyone have a few minutes to discuss network topology regarding asterisk behind a firewall, and two sip trunk carriers. please send me a message. thanks! |
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| 16:40.39 | Kobaz | cmendes0101: call center |
| 16:40.41 | [TK]D-Fender | ipn: Just ask your question. |
| 16:43.10 | [TK]D-Fender | Kobaz: That really doesn't explain much |
| 16:43.33 | [TK]D-Fender | Kobaz: What phone out there can you NOT end calls with? |
| 16:43.41 | ipn | I have a /29 to use for the network. My immediate go-to set-up will be to put the asterisk server on a public IP off the /29 and harden the IPtables along with other monitoring scripts and lock down methods. Then add the router on a different IP off the same /29 |
| 16:43.50 | ipn | and have all the phones register through the router to the public asterisk server and limit only registrations from that router's IP address. |
| 16:44.07 | cmendes0101 | Kobaz: If they hangup just have it ring right back :) |
| 16:44.20 | ipn | I then would add the three trunks I need such as inbound/outbound, international, and 911 to the asterisk box |
| 16:44.31 | ipn | However, I do think this is best practices. It is my understanding to move the asterisk box behind a router/firewall and have the phones on the same subnet of the asterisk box. Then the router/firewall will do the trunking to the vendors. |
| 16:44.40 | ipn | I dont know which is best nor do I know the hardware for the router/firewall device. |
| 16:45.39 | Kobaz | cmendes0101: yeah, heh |
| 16:45.58 | Kobaz | [TK]D-Fender: i've seen call center systems where the agent cannot hang up the call |
| 16:46.10 | Kobaz | yeah the ring back makes sense |
| 16:46.45 | [TK]D-Fender | ipn: That idea works just fine |
| 16:46.59 | [TK]D-Fender | ipn: Security all depends on what you know and how you configure it. |
| 16:47.21 | [TK]D-Fender | Kobaz: No SIP phone would do this that I could think of. |
| 16:47.39 | Kobaz | they had a custom soft phone on the pc |
| 16:47.42 | Kobaz | i remember seeing a demo forever ago |
| 16:47.47 | Kobaz | seems like something this one site could use |
| 16:47.48 | [TK]D-Fender | ipn: As for what you'd want to use as a router.... you don't have any preferencees of your own yet? |
| 16:48.07 | [TK]D-Fender | ipn: I deployed my home setup like you described and am a happy Mikrotik user |
| 16:49.14 | Kobaz | i think this person will just get fired |
| 16:49.15 | Kobaz | but |
| 16:49.23 | Kobaz | they pick up queue calls and then hang up on them |
| 16:49.43 | [TK]D-Fender | ipn: RV's have been known to have NAT issues... would never be one I'd touch |
| 16:51.16 | [TK]D-Fender | ipn: I own a RB2011UiAS-2HnD-IN for home use |
| 16:51.32 | [TK]D-Fender | I have the RM version at the office as a secondary router, but use a SonicWALL as my primary |
| 16:54.45 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-spvsxjxhiggbgahs) |
| 16:54.45 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 16:55.51 | ipn | [TK]D-Fender: I know the RV042s have had NAT issues but the RV130s seem to be doing well. I will look into the Mikrotik model as well since its tried and true with you |
| 16:56.45 | [TK]D-Fender | ipn: They have a massive feature-set and a seriously agressive price-point |
| 16:57.22 | ipn | [TK]D-Fender: Just so I am clear - you set up your asterisk public facing and put the phones behind the Mikrotik or sonicwall router with the other public IP? |
| 16:58.21 | [TK]D-Fender | At home my server is public on a /29, my phones on a private subnet which the router SNAT's (for general outside) and routes direct locally |
| 16:59.36 | ipn | [TK]D-Fender: right your phones are private IP but the router for your phones use a second IP from the /29 |
| 17:00.06 | [TK]D-Fender | They WOULD... except that the server IP is DIRECTLY routable, so they are not actually NAT in-between |
| 17:00.16 | [TK]D-Fender | The server only has that public IP however |
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| 17:45.09 | Spengler | Hi does anyone use voip phones over VPN? |
| 17:45.36 | Spengler | if so are there any settings within asterisk that will make the performance better? |
| 17:46.45 | [TK]D-Fender | obviously not |
| 17:46.50 | [TK]D-Fender | packets are packets |
| 17:47.01 | [TK]D-Fender | And they are already UN-VPN'd by the time they arrive |
| 17:47.08 | [TK]D-Fender | There is no magic |
| 17:47.47 | WIMPy | Well, you easily shoot yourself in the foot with NAT support on either side. |
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| 17:50.53 | Spengler | I have been testing over a vpn and sometimes I find the quality to be good while other times it is bad. |
| 17:51.26 | Spengler | I was just wondering if changing the codec would be better |
| 17:51.45 | [TK]D-Fender | If it's bad... then packets aren't arrigin in a timely manner |
| 17:52.17 | [TK]D-Fender | Unless you are actually choking out on raw bandwidth then changing codecs will get you low aulity in another codec |
| 17:52.55 | Spengler | bandwidth seems to be okay ; 10Mbps on one line and 50 on the other |
| 17:56.03 | [TK]D-Fender | Check your router performance |
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| 19:53.12 | PoWeRKiLL | how can I force asterisk not to reinvite a sip peer even if he know home |
| 19:53.37 | [TK]D-Fender | Give it a reason to sit in the middle |
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| 20:14.46 | PoWeRKiLL | sorry I didn't explain myself on asterisk 1.4 we have canreinvite=no that will for asterisk not to send invite but it's seems that this change on asterisk 13 |
| 20:19.03 | [TK]D-Fender | directmedia=no |
| 20:21.22 | PoWeRKiLL | it's replace canreinvite ? |
| 20:22.54 | [TK]D-Fender | since 1.6 |
| 20:23.02 | [TK]D-Fender | 1.4 = ANCIENT |
| 20:23.27 | PoWeRKiLL | perfect I'm upgrading a Debian Lenny with * 1.4 to Jessie * 13 :D |
| 20:23.50 | PoWeRKiLL | insecure=port,invite still exist right ? |
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| 20:24.32 | [TK]D-Fender | Yes |
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| 23:03.00 | sweettea | mwi keeps blinking when there are no messages on phone |
| 23:03.23 | sweettea | what should i be ensuring is configured correctly |
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| 23:03.55 | sweettea | i only have 1 vm configured |
| 23:04.35 | sweettea | mwi_from=asterisk enabled in sip.conf |
| 23:04.49 | newtonr | check the folder where the voicemail is stored to see if there is actually nothing |
| 23:05.00 | sweettea | 1 voicemail users configured. |
| 23:05.07 | newtonr | check the SIP messaging to see what Asterisk is indicating to the phone |
| 23:05.09 | sweettea | yea via cli, there is nothing |
| 23:05.22 | sweettea | looking in folder now |
| 23:05.34 | newtonr | "there is nothing" ? |
| 23:06.32 | sweettea | voicemail show users |
| 23:06.40 | sweettea | NewMsg = 0 |
| 23:06.47 | sweettea | the INBOX is empty as well |
| 23:06.57 | newtonr | What about the SIP messaging, what is in the NOTIFY ? |
| 23:07.09 | sweettea | sorry I do not know how one checks that |
| 23:07.17 | newtonr | Ah, what SIP channel driver are you using? |
| 23:07.25 | sweettea | oh PJSIP? |
| 23:07.30 | newtonr | "sip set debug on" or "pjsip set logger on" then wait until you see a NOTIFY going out to the phone |
| 23:07.56 | newtonr | copy paste that NOTIFY message here (sanitize any private info) |
| 23:08.06 | newtonr | err copy paste it to Pastebin or hastebin |
| 23:08.07 | sweettea | dumb question |
| 23:08.13 | sweettea | how do i know if im really using pjsip |
| 23:08.38 | *** join/#asterisk amonk (~amonk@unaffiliated/amonk) |
| 23:09.11 | newtonr | "module show like chan_sip.so" and "module show like chan_pjsip.so" to see what is loaded or running |
| 23:09.46 | newtonr | also, sip.conf is for chan_sip, pjsip.conf is for chan_pjsip |
| 23:10.00 | sweettea | interesting, i am using chan_sip.so then |
| 23:10.35 | WIMPy | have you even configured a mailbox= for that phone in your sip.conf? |
| 23:10.49 | sweettea | Module Description Use Count Status Support Level |
| 23:10.52 | sweettea | chan_pjsip.so PJSIP Channel Driver 0 Running core |
| 23:12.35 | sweettea | WIMPy: indeed |
| 23:12.52 | sweettea | mailbox=extension@default set |
| 23:13.13 | WIMPy | Maybe the message has been lost. |
| 23:14.01 | sweettea | I copied over the entire VM directory from an old asterisk build |
| 23:14.40 | sweettea | i wanted to preserve the stupid busy greet etc messages |
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| 23:25.46 | sweettea | newtonr: I do not see any NOTIFY calls, only sip destructions and registrations |
| 23:25.59 | sweettea | now someone called so the cli went nuts |
| 23:26.07 | sweettea | perhaps the phones are malfuctioning |
| 23:26.15 | sweettea | not due to asterisk's doing |
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