| 00:06.05 | GTXComm | Anyone here do hosted PBX? |
| 00:06.24 | GTXComm | From a carrier perspective? |
| 00:07.37 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 00:10.48 | p3nguin | gtxcomm: I've been meaning to call you, but every time I think of it, it's after business hours. Everything pretty much working out now, or do we still need to review your system? |
| 00:18.43 | aschneiderg | Well, let me thank all the 204 of you. |
| 00:18.51 | *** part/#asterisk aschneiderg (~aschneide@186.29.245.172) |
| 00:23.20 | *** join/#asterisk coppice (~chatzilla@m180-219-222-27.smartone-vodafone.com) |
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| 01:15.54 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 01:17.17 | sawgood | any help/advice to solve this concern would be appreciated: I have outgoing SIP trunk calls showing up at the destination caller ID box showing the telephone number as: UNKNOWN CALLER there is no CNAM just the calling number ... |
| 01:17.52 | sawgood | I spoke to the ITSP and they say they are 'receiving' the number from me as, "asterisk", so they strip it off and use UNKNOWN CALLER as the phone number and no CNAM |
| 01:18.39 | sawgood | from the Asterisk CLI, I see the CallerID(num) leaving correctly as the DID number ... not "asterisk" |
| 01:19.19 | p3nguin | "asterisk" is not a caller ID number. |
| 01:19.36 | sawgood | exactly ... but that is what the carrier is saying they are receiving from my server |
| 01:19.40 | p3nguin | Check your CALLERID(num) or CALLERID(all) to see what's on the line. |
| 01:19.49 | p3nguin | They probably are. |
| 01:19.58 | sawgood | CALLERID(num) is correctly leaving as it should |
| 01:20.24 | sawgood | I do have fail2ban installed .... |
| 01:20.39 | p3nguin | You can transmit caller ID name (CNAM) all you want, but the chances of it actually reaching your called destination are slim to none. |
| 01:20.49 | sawgood | 100% agreed ... |
| 01:21.19 | sawgood | They are telling me the number they are receving is the 'word' asterisk in their call logs |
| 01:22.04 | sawgood | I think I will make a SIP trunk to SIP trunk call (by pass the ITSP) and see what shows up |
| 01:22.49 | p3nguin | ~trunk |
| 01:22.50 | infobot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 01:23.53 | sawgood | why is it called, "SIP trunking" then? |
| 01:23.55 | p3nguin | Find out what CALLERID(all) says on your outgoing line. Verbose(${CALLERID(all)}) |
| 01:23.59 | p3nguin | It's not. |
| 01:24.53 | p3nguin | There is no such thing as SIP trunking, so nothing is accurately and appropriately called SIP trunking. |
| 01:27.47 | p3nguin | If you put Verbose(${CALLERID(all)}) before your Dial() command, you can report what is actually being introduced to the call. |
| 01:28.18 | sawgood | setting that up now |
| 01:29.12 | p3nguin | I have a feeling that you aren't sending appropriate caller ID info. |
| 01:30.35 | sawgood | does the Verbose command need to have something like _X. in front of it? |
| 01:36.05 | sawgood | p3nguin: thank you so much, sir! |
| 01:36.11 | sawgood | you were 100% correct! |
| 01:36.25 | sawgood | it is working now ... |
| 01:38.01 | sawgood | absolutely perfect caller number and CNAM now! |
| 01:39.42 | Nombrandue | anyone know of a good resource for troubleshooting RTP? |
| 01:40.26 | JerJer | Nombrandue: depends |
| 01:40.40 | JerJer | ask a specific question, if someone knows the answer they should chime in |
| 01:42.09 | Nombrandue | Incoming calls from my VOIP provider do not have any audio (both ways, just dead silence) but outgoing through the same provider works perfectly well. From the * box and through both sides of the firewall I see the traffic passing, no NAT's or anything, just direct to the box, but I see the VOIP provider kicking back unreachables for the port |
| 01:42.34 | Nombrandue | SIP shows the ports negotiated and using the ports advertised in the call set up, and I see * walking it's dial plan as well |
| 01:43.27 | Nombrandue | through my SPA to the POTS line works perfectly in and out as well |
| 01:47.35 | JerJer | which direction is the unreachable messages ? ie is your ISP blocking? |
| 01:48.02 | Nombrandue | coming from the providers server |
| 01:48.08 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 01:48.13 | Nombrandue | the problem is I am not sure if the ISP is blocking it or not |
| 01:48.45 | Nombrandue | I could find out, but I don't know of any tools that can help me with UDP |
| 01:49.10 | JerJer | netcat |
| 01:53.16 | Nombrandue | I am not seeing anything that looks like it is being blocked |
| 01:54.53 | Nombrandue | though if they were blocking ports, wouldn't it be seen bidirectionally, not just inbound? |
| 01:56.52 | p3nguin | They can block in just one direction. |
| 01:58.10 | Nombrandue | understandable, but again, since UDP is sessionless wouldn't it be all one way, not just for incoming calls, as with outbound RTP packets are swapped in/out during the call, the SIP inition on 5060 is just from a different end of the connection |
| 02:00.03 | p3nguin | I wasn't following along. |
| 02:00.22 | p3nguin | Did you look at rtp debug on asterisk? |
| 02:00.44 | Nombrandue | yeah, all I get is this: Sent RTP packet to 147.135.32.221:13072 (type 00, seq 017872, ts 018720, len 000160) |
| 02:00.57 | Nombrandue | I can read SIP, just need to learn RTP I guess |
| 02:01.03 | Nombrandue | the sip portion looks solid though |
| 02:03.10 | Nombrandue | I am not getting the 'got' packets, like I do internally: Got RTP packet from 192.168.10.36:23202 (type 00, seq 042393, ts 540941728, len 000160) |
| 02:04.13 | Nombrandue | through both the firewall and the * machine, I get, through a tcpdump, 22:07:56.920977 IP 147.135.20.221 > 192.168.10.9: ICMP 147.135.20.221 udp port 11892 unreachable, length 92 |
| 02:05.33 | p3nguin | So your Asterisk is behind a NAT? |
| 02:05.39 | p3nguin | Did you remember to configure it as such? |
| 02:06.02 | Nombrandue | Did with NAT, and also did with it going directly to the box without any port mangling, both the same results |
| 02:06.18 | p3nguin | Which Asterisk version are you using? |
| 02:06.39 | Nombrandue | 1.6.1.18 |
| 02:07.19 | p3nguin | In your sip.conf, what settings did you configure for it to operate properly behind NAT? |
| 02:08.05 | Nombrandue | nat=on, external ip, local net were set |
| 02:08.51 | p3nguin | I'm not sure if "on" is valid for nat. I'd change it to nat=yes, since that is the typical value. |
| 02:09.07 | p3nguin | I'd have to look in chan_sip to find out, and I'd rather not. |
| 02:10.14 | p3nguin | nat=yes, set values for externip or externhost, set values for localnet, and canreinvite=no |
| 02:10.26 | p3nguin | Run "sip reload" and try a call. |
| 02:10.41 | p3nguin | Well, save the sip.conf changes before sip reload, obviously. |
| 02:10.42 | Nombrandue | did all those, and still dead silence |
| 02:11.02 | Nombrandue | asterisk is reporting what I would expect on a call as well: -- Executing [112@from-broadvoice:8] GotoIf("SIP/sip.broadvoice.com-00000017", "0?who-r-u,1,1") in new stack |
| 02:11.20 | p3nguin | That's the SIP portion of it. |
| 02:11.33 | p3nguin | We know the call is being set up. |
| 02:11.55 | Nombrandue | I also see asterisk trying its best, rtp debug, to send RTP packets |
| 02:12.15 | p3nguin | You've also forwarded UDP port 5060 and whatever UDP port range that's set in rtp.conf? |
| 02:12.36 | Nombrandue | yup |
| 02:12.58 | p3nguin | What about the firewall on the Asterisk system? Did you disable it fully? |
| 02:13.21 | Nombrandue | yes it is fully offline on the asterisk box |
| 02:13.22 | p3nguin | It's behind a NAT and another perimeter firewall, so you probably will be safe without it. |
| 02:14.13 | Nombrandue | I also see the RTP and SIP flow out through the perimeter firewall, and the ICMP Port messages hit both the internal and external interfaces of the firewall and the Asterisk box |
| 02:15.29 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
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| 02:52.28 | Godofmonkeys | i still can't lick this outgoing shot |
| 02:52.30 | Godofmonkeys | shit |
| 02:52.56 | p3nguin | :/ |
| 02:53.01 | Godofmonkeys | fail auth |
| 02:53.17 | Godofmonkeys | can't find it; been through the book a few times now |
| 02:53.58 | Godofmonkeys | still have 5 million other things to config too, voicemail, groups, etc |
| 02:54.13 | p3nguin | I'm not convinced it's your problem, considering the service I set you up with challenges in the same manner that I described. |
| 02:54.18 | Godofmonkeys | beat down the 604 error |
| 02:54.40 | Godofmonkeys | me either, but they are being less than helpful at simpsig |
| 02:54.42 | p3nguin | Your system works with what I gave you, so your system works. |
| 02:54.59 | *** join/#asterisk eugeneoden (~goden@99-62-173-93.lightspeed.austtx.sbcglobal.net) |
| 02:55.12 | Godofmonkeys | no argument there; but if it works, then changing the server and credentials should not break it |
| 02:55.33 | p3nguin | Unless the other server that you're changing _to_ is broken. |
| 02:55.39 | Godofmonkeys | yar |
| 02:55.50 | p3nguin | So I think we have the answer. |
| 02:56.29 | Godofmonkeys | they claim to have many, many users with the same setup as I |
| 02:56.33 | p3nguin | Do they have any customers who don't have problems? |
| 02:56.40 | p3nguin | Then I'd tell them to find another customer who is successfully using Asterisk. |
| 02:57.27 | Godofmonkeys | it's to the point i'm ready to beat this box with a hammer |
| 02:57.50 | p3nguin | I guess I could take your credentials and configure it on my box. If it still doesn't work, then I'd say they are a pile of shit. |
| 02:58.03 | Godofmonkeys | if you don't mind so much |
| 02:58.06 | p3nguin | You want me to do that for a minute? |
| 02:58.14 | Godofmonkeys | hold for pm |
| 02:58.33 | p3nguin | I think I'll just need the server address, user, and pass. |
| 03:21.31 | *** join/#asterisk strDSR (~stardouse@140.144.121.11) |
| 03:31.19 | ruben23 | hi guys where are the voicemail on asterisk is stored..? what directory..? |
| 03:31.23 | ruben23 | default i mean |
| 03:35.11 | p3nguin | /var/spool/asterisk/voicemail/ |
| 03:43.28 | *** join/#asterisk e911_indiana (~medic@c-98-220-104-243.hsd1.in.comcast.net) |
| 03:44.34 | e911_indiana | having an issue with getting asterisk to read one of my extensions on my phone. basically what i did was migrate from one system to another, saved the configs to a safe location, uploaded them to the new server, adjusted settings on the phone, and yet, i can not get just 1 extension to come up, but the rest have come up just fine and can make calls all day long. thoughts? |
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| 03:49.33 | p3nguin | godofmonkeys: Okay, I got it to work. |
| 03:50.39 | p3nguin | godofmonkeys: Do you have a DID where I can test an inbound call to it? |
| 03:53.59 | Godofmonkeys | the user name |
| 03:55.33 | p3nguin | oh |
| 03:55.46 | p3nguin | So the callerID was right, then. |
| 03:56.21 | p3nguin | I wasn't sending any CID number, but MCI still told me that number was the phone number. I didn't put two and two together to realize it was the username. |
| 03:56.37 | Godofmonkeys | we'll have to have our number ported to this at some point, but yes, that's my did |
| 03:57.28 | Godofmonkeys | ... it will probably break shit, but... |
| 03:57.38 | p3nguin | So that number is tied to the account? Meaning if I dial it, the call should come to me? |
| 03:57.59 | *** join/#asterisk OrNix (~ornix@l49-131-248.cn.ru) |
| 03:58.03 | Godofmonkeys | yes, it was hitting my * server, but apparently my incoming dialplan is wrong |
| 03:58.14 | p3nguin | I just didn't want to dial it and it ring your phone. |
| 03:58.38 | Godofmonkeys | no problem |
| 03:59.55 | p3nguin | Inbound calls don't come to me, for an unknown reason. |
| 04:00.12 | p3nguin | I'm registered, so their side should know where to send the call. |
| 04:01.24 | p3nguin | Wait, I think I lied. |
| 04:10.18 | p3nguin | godofmonkeys: Give your number a call. |
| 04:10.25 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ixtlernitkkbadcp) |
| 04:10.34 | Godofmonkeys | ok, one moment |
| 04:10.41 | p3nguin | I've got it all done. |
| 04:11.21 | Godofmonkeys | echo echo echo..... |
| 04:11.33 | Godofmonkeys | that's beautiful |
| 04:11.55 | Godofmonkeys | so how far off was I? |
| 04:12.35 | p3nguin | I have no idea. I started with the raw account info that you gave me and built up a peer definition until I got the right settings. |
| 04:13.50 | Godofmonkeys | hm, i'll have to pull a diff and see what I did wrong |
| 04:14.19 | p3nguin | The good news is that their system does work with Asterisk. |
| 04:14.34 | p3nguin | two-way calling, even. |
| 04:15.30 | Godofmonkeys | well good, now i just have to make mine work with it |
| 04:15.44 | p3nguin | I have the config. |
| 04:21.38 | Godofmonkeys | well what will you have me do? |
| 04:23.52 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:26.03 | e911_indiana | well hellz bellz.. that didnt work :( time to reflash the phone.. |
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| 05:26.04 | neurosys | Why would an external server display "SIP read from UDP:192.168.1.200:5060 --->" in the debug? Im trying to connect 2 boxes via SIP, but with nat enabled, its reading as if its coming from the local IP of the other network |
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| 05:32.58 | Schreiber1337 | Getting soo fustrated with Call Pickup... anyone know what I would get this "pickup_exec: No target channel found" for a ringing phone? |
| 05:33.24 | neurosys | are they in different contexts? |
| 05:34.14 | Schreiber1337 | Nope... calling phone, called phone, and the phone that is trying to pickup all are using [internal] context |
| 05:34.39 | Schreiber1337 | Here is an example... http://dl.dropbox.com/u/7097983/3.txt.txt |
| 05:34.42 | neurosys | pastebin the contect. |
| 05:34.45 | neurosys | heh k |
| 05:35.16 | neurosys | does the incoming call come in from another context? |
| 05:36.11 | Schreiber1337 | No... I'm just trying all from within the [internal] context |
| 05:36.57 | Schreiber1337 | Hmmm... wait a minute... |
| 05:37.56 | Schreiber1337 | Nope... [unrestrictedInternational] has an include for [internal] |
| 05:38.09 | Schreiber1337 | So... it should work |
| 05:38.13 | godofmonkeys | Success! Incoming and outgoing. Thanks again p3nguin. |
| 05:39.18 | neurosys | Pastebin extensions.conf? |
| 05:40.14 | Schreiber1337 | neurosys: http://dl.dropbox.com/u/7097983/4.txt |
| 05:48.31 | Schreiber1337 | neurosys: That worked... Thanks! Now if I can just get this Cisco phone to send the pickup string correctly... I'll be golden... Thanks again! |
| 05:48.52 | neurosys | :) |
| 05:48.57 | neurosys | Glad I could help |
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| 06:45.45 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:45.48 | schmidts | good morning |
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| 06:47.09 | Dryanta | hy |
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| 06:49.44 | Dovid | morning |
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| 06:54.57 | schmidts | any kamailio or openser geeks here? |
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| 07:02.22 | Dovid | I use OpenSipS but by no means an OpenSipS geek. whats up ? |
| 07:03.02 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:8d14:cd63:25af:533b) |
| 07:05.06 | schmidts | something really stupid and wierd problem ;) |
| 07:06.04 | schmidts | we have many of linksys / Cisco Spa phones and using subscribe for BLF functions. After setting up a proxy with kamailio in front of our asterisk, this stupid phones just send the first subscribe to the proxy and the resubscribe directly to asterisk |
| 07:07.30 | *** join/#asterisk Godofmonkeys (~Godofmonk@71-87-181-173.dhcp.jcsn.tn.charter.com) |
| 07:07.37 | schmidts | i have allready set up in kamailio that i change the contact header for subscribes and notifys, removed the ip of asterisk and just set the proxy ip in it but the 200 OK message for the first resubscribe still isnt changed, so after the first resubscribe the phone again sends the second subscribe to asterisk instead of the proxy |
| 07:08.08 | Godofmonkeys | hey guys, where would one find info on polycom phone conifguratons |
| 07:09.24 | Dovid | schmidts: I spaced out for this part of the course ;). Maybe you want to insert a recordroute(); so your OpenSipS sits in the path ? |
| 07:10.27 | Dovid | schmidts: Can you pb a trace? also how do you change the headers ? I have been looking to do that for some other things |
| 07:10.33 | schmidts | dovid i allready sets the recordroute even for indialiog pakets ;) |
| 07:10.39 | schmidts | yes mom |
| 07:10.46 | Dovid | Godofmonkeys: Polycom.com ;). What do you need to know? |
| 07:11.02 | Godofmonkeys | configuration from scratch |
| 07:12.19 | Dovid | Godofmonkeys: PM me in an hour. I will be in the office and can look at my phone there |
| 07:12.38 | Dovid | Godofmonkeys: In Asteisk source you should have some example there |
| 07:12.42 | Godofmonkeys | ok, i'm gonna continue to read up on them. |
| 07:14.02 | Dovid | there are examples in the phoneprov directory |
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| 07:15.07 | schmidts | dovid http://pastebin.com/hcQwczkf |
| 07:15.31 | schmidts | dovid thats a trace from the first resubscribe and the "wrong" 200 ok |
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| 07:37.06 | Maxus2 | Hi asterisk people, does anyone know why i'm not seeing CDR records for outbound calls? |
| 07:37.28 | schmidts | dovid ok i found the problem i didnt change the contact for in dialog subscribe requests :( |
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| 07:47.01 | kaldemar | Maxus2: have you disabled them in your dialplan? |
| 07:50.01 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 07:54.34 | Maxus2 | no i dont think so. |
| 07:54.44 | Maxus2 | my dial plan: |
| 07:55.03 | Maxus2 | [to-client] |
| 07:55.04 | Maxus2 | exten => _089511XXXX, 1, Dial(SIP/101) |
| 07:55.04 | Maxus2 | exten => _089511XXXX, n, Hangup |
| 07:55.04 | Maxus2 | [from-client] |
| 07:55.04 | Maxus2 | exten => _., 1, Set(CALLERID(name)=PBX One) |
| 07:55.04 | Maxus2 | exten => _., 2, Dial(IAX2/RCOM-000001-0001/${EXTEN}) |
| 07:55.05 | Maxus2 | exten => _., n, Hangup |
| 07:56.42 | Maxus2 | according to this: https://issues.asterisk.org/view.php?id=2246 there is no outbound cdr entry created? |
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| 08:03.20 | kaldemar | Maxus2: are you getting any kind of entry for the call? |
| 08:09.30 | Maxus2 | yep i get the inbound record |
| 08:09.38 | Maxus2 | but not an outbound |
| 08:09.48 | Maxus2 | im passing the call to another asterisk box |
| 08:09.53 | *** join/#asterisk sekil (~sekil@78.24.111.154) |
| 08:10.35 | kaldemar | why do you want a separate record for the outbound leg? |
| 08:12.48 | Maxus2 | billing |
| 08:13.46 | kaldemar | you can use the record that you have. there's a field for the destination channel. |
| 08:14.07 | Maxus2 | yeah just noticed the last data field |
| 08:14.09 | kaldemar | but if you absolutely want to, you can create a new record with application ForkCDR. |
| 08:14.14 | Maxus2 | just seems a bit messy |
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| 08:16.21 | Dovid | Back |
| 08:17.44 | Maxus2 | cool thanks kaldemar, thats what i thought i was hoping for somthing nicer. |
| 08:17.51 | Maxus2 | hopeing |
| 08:21.25 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:28.08 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 08:28.21 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 08:36.20 | Godofmonkeys | having some trouble with voicemail. I have it where i can call VoiceMailMain, but an incoming call never goes to the vm prompt |
| 08:39.13 | kaldemar | what do you see in CLI with verbosity and core debug enabled? |
| 08:39.38 | Godofmonkeys | have not turned it on |
| 08:39.42 | Godofmonkeys | just a sec |
| 08:41.18 | Godofmonkeys | sorry, i know the sip debug commands... |
| 08:41.43 | kaldemar | core set verbose 10, core set debug 10 for example. |
| 08:42.17 | Godofmonkeys | one moment, it seems i'm being flooded with register requests |
| 08:42.36 | kaldemar | set sip debug off so it won't bother you. |
| 08:42.46 | Godofmonkeys | it is, there notice lines |
| 08:43.05 | Godofmonkeys | random numbers from a strange ip |
| 08:43.57 | Godofmonkeys | makes cli unusable |
| 08:44.53 | kaldemar | get the output to a file with script or some other tool, and then filter the unwanted lines. |
| 08:46.51 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:47.08 | Godofmonkeys | is there a way to just block his ip? |
| 08:48.52 | kaldemar | iptables -I INPUT -p all -s ip.address -j DROP |
| 08:50.11 | Godofmonkeys | thank you. |
| 08:51.10 | kaldemar | if you get hammered a lot with register requests, you might want to install fail2ban to block attackers automatically. |
| 08:51.30 | Godofmonkeys | i will look into it |
| 08:51.42 | Godofmonkeys | it's been up less than 8 hours |
| 08:52.11 | Godofmonkeys | looks like he was tring to brute force an extension he apparently discovered |
| 08:53.17 | kaldemar | this is also a very good read: http://svn.digium.com/svn/asterisk/tags/1.8.3.2/README-SERIOUSLY.bestpractices.txt |
| 08:55.54 | Godofmonkeys | kaldemar, this is what i get from the cli: http://pastebin.com/AxYJ3yKP |
| 08:57.38 | Godofmonkeys | extensions.conf: http://pastebin.com/xL2cWhpr |
| 09:00.52 | kaldemar | well, you're dialing a number that has anything to do with voicemail in your dialplan. |
| 09:01.00 | kaldemar | s/anything/nothing/ |
| 09:02.27 | Godofmonkeys | that's how it's laid out in the wiki.asterisk.org page on configuring voice mail boxes |
| 09:03.41 | kaldemar | your call goes to 3038000142@simpsig-inbound:1. the extension has no voicemail application or a way to go to an extension that does. |
| 09:03.49 | kaldemar | what do you want to do? |
| 09:04.25 | Godofmonkeys | does that not happen when it gets passed to 100 via the Dial(SIP/100)? |
| 09:05.10 | kaldemar | it doesn't get passed anywhere, it dials SIP/100 which means device [100] in sip.conf. |
| 09:06.06 | Godofmonkeys | so as it is now, i would only get a voicemail prompt from a 2nd extension? |
| 09:06.15 | kaldemar | if you want to go to another extension in your dialplan, you need to use app Goto or dial a local channel (Dial(Local/exten@context)). |
| 09:07.00 | kaldemar | 300@myphones has VoiceMailMain() |
| 09:07.37 | Godofmonkeys | right; i can dial 300 to check the mail. it asks for a mailbox number and password |
| 09:09.06 | kaldemar | replace "exten => 3038000142,n,Dial(SIP/100)" with "exten => 3038000142,n,Goto(myphones,100,1)" if you want the incoming call to land in that extension. |
| 09:10.39 | Godofmonkeys | and as far as what i want to do; the list is long. as it stands, i can call in and out. i'm using x-lite to test with, untill i setup auto provision for my polycom 321. I need to setup an IVR to catch all incoming, play a nice recording, give some press 1 for directions, 2 for hours, 3 to actually speak to a person... then it needs to ring a select few at the same time..... there is lots to do |
| 09:11.32 | Godofmonkeys | so i guess i'm working ahead of myself, i need a group.. but need to setup polycom... |
| 09:14.22 | Godofmonkeys | kaldemar, the goto line did the trick. |
| 09:14.28 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 09:15.16 | kaldemar | ~newbook |
| 09:15.16 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 09:15.33 | kaldemar | have a look at dialplans in there ^ |
| 09:16.04 | Godofmonkeys | heh, i have the book bookmarked :) |
| 09:20.08 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.229.178) |
| 09:21.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
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| 09:41.16 | *** join/#asterisk flashdeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 09:42.05 | flashdeluxe | Hi! Is there anybody who knows a good monitoring tool for * ? |
| 09:45.54 | ruyo | asterisk -r |
| 09:45.58 | ruyo | :) |
| 09:47.38 | flashdeluxe | for several servers in different locations ;) |
| 09:47.56 | *** join/#asterisk _zoom_ (~Admin@196.1.219.211) |
| 09:48.59 | _zoom_ | hello, when you have two matched entries in the dialplan, does it execute the both exten or only the first one? |
| 09:53.17 | kaldemar | it executes the better match in case of extensions in the same context. |
| 10:02.29 | _zoom_ | is there any var to set to execute them all sequenctially ? |
| 10:09.53 | kaldemar | no. if you want more things done in your dialplan, put them in the extension that the call goes to. or into another extension and execute it with Gosub. |
| 10:13.15 | *** part/#asterisk sekil (~sekil@78.24.111.154) |
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| 11:46.04 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
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| 11:56.22 | *** part/#asterisk freckle (~freckle@87.127.248.44) |
| 12:12.49 | *** join/#asterisk Kako (~Kako@213.240.177.207) |
| 12:15.05 | *** join/#asterisk obelix (~kvirc@unaffiliated/pavlz) |
| 12:15.10 | obelix | hello |
| 12:15.16 | Kako | Hi, Im using a asterisk 1.6.2.16.1. I noticed that the nr of international incomming calls aren't passed correctly on the the internatl SIP. The leading 00 is trimmed to a single 0. On national calls the nr stays like it is. Anyone an idea why the leading 00 ist trimmed and how I can change it? |
| 12:15.19 | obelix | one query |
| 12:15.38 | obelix | can i use asterisk as a program for call center ? |
| 12:16.11 | *** join/#asterisk mawhii (~mawhii@170.220.119.70.cfl.res.rr.com) |
| 12:16.19 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:16.47 | Kako | obelix: It depend on what asterisk should do on the call-center. It offers for example call-queues. I woul d say: yes, you can |
| 12:20.19 | kaldemar | Kako: how are your calls coming in? |
| 12:21.36 | Kako | kaldemar: the calls are commin in over a dialogic Diva PRI/E1/T1-o PCI Card (ETSI-DSS1) |
| 12:21.38 | obelix | for inbound and outbound calling |
| 12:22.38 | Kako | obelix: Sorry, what d you mean? Asterisk is a PBX so it can also manage inbound and outbound calls. |
| 12:23.12 | Kako | Maybe you can rewrite your question |
| 12:23.44 | kaldemar | Kako: see pri debug to find out what your provider sends. |
| 12:23.44 | benngard | Kako: howdo u receive batinal calls? witout a leading 0 or? |
| 12:23.54 | benngard | national calls* |
| 12:25.02 | Kako | kaldemar: I must receive the full nr from my provider, because I can see it inthe asterisk-log "...chan_capi_utils.c: == ISDN2#29: Incoming call '00..." |
| 12:26.19 | Kako | benngard: national calls I see in the log as 08154711, international calls 004908154711 |
| 12:26.51 | Kako | benngard: (fix) international calls 0498154711 |
| 12:26.57 | Kako | benngard: (fix2) international calls 00498154711 |
| 12:26.59 | Kako | :-) |
| 12:27.26 | kaldemar | capi.. hmm... there's a pridialplan setting in dahdi channel config that might affect, but don't know about capi. |
| 12:28.38 | kaldemar | if you can't fix it in the channel driver, you can do what you want to all numbers in dialplan though. |
| 12:31.58 | *** join/#asterisk josta (~josta@unaffiliated/josta) |
| 12:32.42 | *** join/#asterisk coppice (~chatzilla@m180-219-222-27.smartone-vodafone.com) |
| 12:33.11 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:35.22 | Kako | kaldemar: all options in chan_dahdi.conf seems to set to default values or commented aout. Any suggestion? Setting this over dialplans loks for me like a workaround without fixing the problem an would unnecessary increase the nr of rules |
| 12:42.23 | kaldemar | Kako: you seem to be using capi. i never have so i don't know. |
| 12:43.50 | Kako | kaldemar: It's a pity... yes, I'm using capi. But thanks for you spended time. |
| 12:44.40 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
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| 12:47.06 | kaldemar | Kako: there seem to be all sorts of options in the capi.conf regarding this. nationalprefix, internationalprefix, prefix... |
| 12:48.03 | Kako | kaldemar: yes, but the seem to be correct: |
| 12:48.04 | Kako | nationalprefix=0 |
| 12:48.06 | Kako | internationalprefix=00 |
| 13:00.03 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 13:05.29 | schmidts | can anyone help me with Cisco 7490 phones? I have a problem that they cant talk anymore after i put a kamailio proxy in front of my asterisk. Sip signaling works fine but RTP didnt work anymore, cause this should go to another server (asterisk) |
| 13:06.16 | schmidts | thats my config for the phones: http://pastebin.com/AcdiAnp9 |
| 13:12.05 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 13:15.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:20.52 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 13:21.29 | Kobaz | i have two boxes where at 6-7am every morning, all the polycom phones and pcs lose their ip address |
| 13:21.32 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:21.53 | Kobaz | anyone else ever have a problem like that? |
| 13:21.59 | Kobaz | my dhcp lease time is like two weeks |
| 13:22.20 | guilhermebr | what do you say about enable sip jbuffer ? |
| 13:22.31 | n3hxs | Is the janitor unplugging the machine to run the floor polisher? |
| 13:23.26 | Kobaz | hah, no |
| 13:23.35 | n3hxs | just a wild guess ;) |
| 13:23.37 | Kobaz | the phones are acting really weird with dhcp too |
| 13:23.45 | Kobaz | like, even though it's a really long lease |
| 13:23.56 | Kobaz | the phones are doing a dhcpdiscover every minute or so |
| 13:24.24 | Kobaz | er, not discover, a dhcprequest |
| 13:26.13 | Kobaz | i think i have to diff my config vs a good system |
| 13:26.29 | Kobaz | eyeballing it, the dhcp config looks exactly the same as a good system |
| 13:26.49 | Kobaz | maybe i should clear the lease file, maybe it's got some screwy data |
| 13:32.09 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
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| 13:44.48 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 13:46.36 | iprouteth0 | anyone know what the http compile option does in 1.8? |
| 13:58.04 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
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| 14:06.05 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:06.07 | Aut0ExeC | I have a really dumb question.... if I have a voip number example 555-555-5555 and I get an incoming call... and transfer the call... I cant receive another call on that line until the person I transfered the call to has hung up right? |
| 14:07.40 | jaytee | not necessarily |
| 14:07.50 | iprouteth0 | depends on how many channels you have on the trunk |
| 14:07.55 | *** join/#asterisk rgenthner (~rgenthner@cpe-74-75-158-14.maine.res.rr.com) |
| 14:09.00 | jaytee | I have multiple DIDs from flowroute and I can get multiple concurrent calls on a single DID when I choose the metered option vs the unlimited option. The unlimited option has a fixed cost per month but limits me to 2 concurrent calls |
| 14:09.41 | Aut0ExeC | ahhh i see |
| 14:09.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:09.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.47 | Aut0ExeC | so that is setup on the provider right? |
| 14:09.52 | iprouteth0 | correct |
| 14:09.56 | Aut0ExeC | ok great |
| 14:09.57 | jaytee | yes, it depends on your provider |
| 14:10.07 | iprouteth0 | some have the option of adding more channels per DID |
| 14:10.28 | Aut0ExeC | so the more channels = more concurrent calls? |
| 14:10.38 | *** join/#asterisk X-Raimo (~X-Raimo@opensuse/member/raimoschmidt) |
| 14:11.03 | rgenthner | anyone use qoutcon with realtime mysql queue setup ? |
| 14:11.06 | iprouteth0 | yes. Though it might be called different things by different providers |
| 14:11.16 | Aut0ExeC | iprouteth0: ok sounds hot man.. thanx |
| 14:11.22 | X-Raimo | hi all. How to disable caching user and password for specific realtime peer? |
| 14:12.55 | iprouteth0 | Aut0ExeC: np |
| 14:16.05 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:17.03 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:18.54 | Aut0ExeC | iprouteth0: you have a digium card? |
| 14:19.02 | Aut0ExeC | iprouteth0: 24 channel big boi? |
| 14:20.28 | iprouteth0 | Aut0ExeC: nah, i wish. I have a polycom IP450 and linksys PAP2 that I work with |
| 14:23.12 | *** join/#asterisk neurosys (~neurosys@173.200.196.25) |
| 14:23.19 | *** join/#asterisk breardo (~breardo@234-200-29-134.hcc.mnscu.edu) |
| 14:23.25 | breardo | hello all |
| 14:23.32 | *** join/#asterisk sethbourne (seth_20@118.96.2.40) |
| 14:23.45 | breardo | i have a 1.6.17.2 box that, approximately every 30 minutes, all the sip peers become unqualified, and then requalified.. |
| 14:23.57 | breardo | it results in temporary one-way-audio and other audio-issues during a call |
| 14:24.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:24.40 | sethbourne | hi there |
| 14:24.48 | breardo | hi seth |
| 14:25.02 | breardo | sethbourne sounds like a nice place to live :) |
| 14:25.08 | breardo | i like the way that sounds |
| 14:25.14 | sethbourne | haha.. thank you |
| 14:25.36 | sethbourne | I need some information about publish asterisk behind nat, and access client behind nat. any1 have a documentation about that ? |
| 14:25.54 | breardo | thats generally problematic, from what I know of it.. |
| 14:25.58 | breardo | but thats about all :) |
| 14:26.20 | breardo | but do-able, and im sure someone in here can address the question better than I :) |
| 14:26.26 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 14:26.29 | sethbourne | :) |
| 14:27.10 | tuxxie | Has anyone been able to integrate asterisk with MS Dynamics online CRM? |
| 14:29.52 | Aut0ExeC | iprouteth0: thats hot bro... I have a cisco spa3102... doesnt work to well with pstn line tho... doesnt hangup the call when disconnected |
| 14:30.51 | Aut0ExeC | sethbourne: I thought it was just a matter of opening ports |
| 14:31.18 | Aut0ExeC | sethbourne: I can connect to my server by just opening ports 5060 and 10000-20000 |
| 14:31.25 | Aut0ExeC | udp |
| 14:31.38 | iprouteth0 | sethbourne: forward your SIP port, and set externip or externhost |
| 14:32.10 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:32.27 | Moos3__ | does anyone know if theres any bugs related to realtime queues in 1.6 ? |
| 14:35.53 | breardo | anyoen else ever see all their peers become unqualified then requalified every ~30 minutes? |
| 14:36.17 | breardo | the peers are pingable, with >1ms reply times.. |
| 14:36.19 | breardo | no nat |
| 14:36.26 | saxa | hi, does asterisk need some special privs to read a sound file from a users home dir ? |
| 14:36.54 | saxa | my install runs asterisk as root |
| 14:37.06 | serafie | Moos3__: 1.6 isn't an Asterisk version. Can you be more specific? |
| 14:37.15 | saxa | I want to play that sound with Playback() |
| 14:37.39 | breardo | saxa, you may want to try symlinknig to it in your asterisk sounds dir, and then referencing the symlink? |
| 14:37.43 | Moos3__ | 1.6.2.16.1 |
| 14:38.14 | breardo | exit |
| 14:38.16 | breardo | DOH |
| 14:38.18 | breardo | :) |
| 14:38.23 | Aut0ExeC | iprouteth0: hey u still ther bro? |
| 14:38.57 | serafie | Moos3__: https://issues.asterisk.org/view.php?id=15967 affects 1.6.2.0 |
| 14:39.01 | iprouteth0 | yep |
| 14:39.11 | Moos3__ | serafie: this makes no sense why my realtiem queues are ignoring keypress exit to context |
| 14:39.16 | Aut0ExeC | iprouteth0: how do you configure your poly? |
| 14:39.24 | Aut0ExeC | iprouteth0: i have never setup an ip phone |
| 14:39.48 | saxa | breardo: just specify the full path isnt enough ? |
| 14:40.00 | saxa | breardo: will try to symlink it |
| 14:40.35 | iprouteth0 | well... You can do so with config files that the phone will load from http/ftp/tftp server, or what I did was just configure it with web GUI since I only have one to worry about |
| 14:40.35 | serafie | Moos3__: that doesn't sound like it would be affected by realtime. |
| 14:41.12 | zkn | i've got two servers, an old and a new one, different machines located in different countires. now, i want to move the trunk account from the old server to the new server, but even though I get the SIP trunks registered on the new server, when I call the DIDs then the only thing appearing to CLI is:" == Using SIP RTP CoS mark 5" and the call ends with busy signal, what could be the issue? |
| 14:41.21 | iprouteth0 | asterisk has a zeroconf feature and also a polycom autoprovision feature I've yet to use |
| 14:41.47 | Aut0ExeC | iprouteth0: ok nice man |
| 14:41.54 | Moos3__ | serafie: no matter what happens when i'm on holding waiting for a agent, it fails to see that i pressed one, it does pause the music for the keypress but thats it |
| 14:41.57 | Aut0ExeC | iprouteth0: i think i'll get one of those |
| 14:43.04 | iprouteth0 | they're not the cheapest, but they are nice phones. |
| 14:43.09 | Aut0ExeC | ok |
| 14:43.20 | iprouteth0 | I saw another cool one I would love to try though you might want to consider |
| 14:43.36 | Aut0ExeC | whats that? |
| 14:43.44 | Aut0ExeC | cisco? |
| 14:43.52 | iprouteth0 | nortel 1535 |
| 14:43.59 | Aut0ExeC | ok |
| 14:44.00 | *** join/#asterisk e911_indiana (~medic@c-98-220-104-243.hsd1.in.comcast.net) |
| 14:44.11 | breardo | saxa.. i think asterisk uses chroot, so it wouldnt 'see' the users home dir.. |
| 14:44.17 | iprouteth0 | I've been seeing them around 70$ for non wifi |
| 14:44.23 | iprouteth0 | video SIP |
| 14:44.48 | serafie | Moos3__: pastebin the exit context |
| 14:45.04 | Moos3__ | k |
| 14:45.04 | Aut0ExeC | wow nice man.. |
| 14:45.05 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:45.58 | serafie | also queues.conf |
| 14:46.09 | Moos3__ | serafie: https://gist.github.com/e5880ff8da972c259c7b |
| 14:46.19 | Moos3__ | serafie: my queues are real time |
| 14:46.23 | saxa | breardo: symlinking it worked out |
| 14:46.26 | saxa | thx |
| 14:46.31 | serafie | Doh, yeah. |
| 14:46.43 | serafie | You've got no "1" extension in your exit context |
| 14:46.54 | serafie | change "quotcon" to "1" |
| 14:47.22 | sethbourne | @iprouteth0: hi there, i've doing just as you said. forward port 5060, 10000-20000, I can connect it, but I cant hear any voice |
| 14:47.22 | breardo | great, glad it helped :) |
| 14:47.28 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:47.51 | breardo | saxa, be warned.. if that referenced file changes often, your symlink could break.. |
| 14:48.24 | Aut0ExeC | sethbourne: u try another phone? like 2 softphones?> |
| 14:48.43 | sethbourne | aut0exec: yes i have |
| 14:48.52 | sethbourne | aut0exec: yes i did |
| 14:49.12 | iprouteth0 | sethbourne: rtp set debug on may help show whats going on |
| 14:49.17 | Moos3__ | serafie: k |
| 14:49.21 | saxa | breardo: its just to test the brasilian sounds, after i have them checked, i will place them in the asterisk sounds dir. |
| 14:49.37 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 14:49.45 | iprouteth0 | sethborne: did you set externip or externhost in sip.conf? |
| 14:50.32 | sethbourne | externip |
| 14:51.04 | Moos3__ | serafie: like this https://gist.github.com/e5880ff8da972c259c7b |
| 14:51.21 | serafie | Yes |
| 14:51.24 | iprouteth0 | rtp debug might be best start |
| 14:52.55 | sethbourne | iprouteth0: where i can run rtp debug ? |
| 14:53.29 | iprouteth0 | sethbourne: in asterisk cli. From a terminal type |
| 14:53.29 | iprouteth0 | asterisk -rvvv |
| 14:53.29 | Moos3__ | serafie: thanks |
| 14:53.42 | *** join/#asterisk slidesinger-lt (~jtatum@12.149.62.10) |
| 14:53.55 | sethbourne | iprouteth0 : thanks |
| 14:54.39 | breardo | seth.. you will want to setup logger.conf and dump relvant output to a file, |
| 14:54.52 | breardo | that rtp and sip debug output gets deep, fast. |
| 14:55.16 | breardo | checkout the HOWTO on collecting debug info in the doc dir in your source.. |
| 14:55.48 | sethbourne | breardo : ok then.. i will try a look |
| 14:56.11 | breardo | its easy, and you wont regret knowing how to capture specific debug output to a file.. |
| 14:56.30 | iprouteth0 | thanks for the tip breardo! |
| 14:56.34 | breardo | especially on an active PBX, holy crap... too much screen output :) |
| 14:56.39 | breardo | no prob iprouteth0 |
| 14:59.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:00.48 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 15:01.46 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:03.14 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:05.30 | breardo | anyone ever see all sip peers unqualify/requalify every 30 minutes, with temporary one-way-audio issues at the same time? |
| 15:05.38 | breardo | unqualify/requalify happens within 2 seconds or so |
| 15:05.47 | breardo | 1.6.2.17.2 |
| 15:09.00 | Aut0ExeC | breardo: bad connection on ur switch? |
| 15:09.14 | Aut0ExeC | breardo: more than 1 peer? |
| 15:09.27 | Aut0ExeC | oh all... just saw that |
| 15:09.33 | Aut0ExeC | weird |
| 15:11.12 | breardo | yeah its all peers.. 47 of them |
| 15:11.25 | breardo | i dont think its a bad connection, as it repeats every ~30 mins.. |
| 15:11.44 | breardo | and i dont see any errors in netstat, or on the switch port .. |
| 15:12.44 | breardo | i agree, it sounds network-related.. |
| 15:15.02 | Aut0ExeC | yup |
| 15:15.58 | Aut0ExeC | tcpdump on the line |
| 15:16.43 | breardo | thats next.. using asterisk to capture data at the moment.. |
| 15:16.49 | breardo | sip and rtp debugs.. |
| 15:17.00 | breardo | it should have happened at 10:10 (or so) but it didnt this time.. |
| 15:17.08 | breardo | throwing the "every 30 minutes" part out the window. |
| 15:17.11 | breardo | damn :) |
| 15:17.31 | breardo | err, it should have happened seven minutes ago :) |
| 15:20.01 | breardo | ahh it just happened |
| 15:20.03 | breardo | 8 minutes late |
| 15:26.36 | *** join/#asterisk m_tadeu (~quassel@89-180-134-250.net.novis.pt) |
| 15:27.42 | ChannelZ | PoE? |
| 15:27.44 | *** join/#asterisk voila (IceChat77@117.212.65.182) |
| 15:27.49 | voila | hello |
| 15:27.52 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 15:27.54 | m_tadeu | hi...when configuring extconfig.conf, what's the difference between sippeers and sipusers? |
| 15:28.30 | voila | hey .. can any one explain me .. how can i use dial commmnd to dial a softphone ?? i gooogle it , but found very encrypted things |
| 15:28.38 | breardo | no.. no PoE |
| 15:29.08 | ChannelZ | voila: Dial(SIP/SoftPhoneName) where SoftPhoneName is whatever device name you gave it in sip.conf |
| 15:29.24 | ChannelZ | (presuming it's a SIP softphone as that's most typical) |
| 15:29.25 | m_tadeu | voila: you may want to check this...http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 15:29.37 | voila | ChannelZ : ok let me check .. |
| 15:29.44 | voila | m_tadeu: ok |
| 15:31.53 | serafie | voila: also https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 15:32.01 | iprouteth0 | google voice integration with 1.8 is sweet! |
| 15:32.31 | breardo | weird.. |
| 15:32.43 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 15:32.46 | breardo | the RTP debug shows perfect send/received and then it just stopps.. |
| 15:32.48 | breardo | err stops |
| 15:32.54 | breardo | and just sent.. sent..sent.. no rcv.. |
| 15:33.18 | breardo | http://pastebin.com/23Dt8ZTJ |
| 15:35.02 | voila | [default1] |
| 15:35.02 | voila | exten => 1234 , 1, Dial(SIP/3005) |
| 15:35.06 | voila | <PROTECTED> |
| 15:35.07 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:35.11 | voila | [default1] |
| 15:35.11 | voila | exten => 1234 , 1, Dial(SIP/3005) |
| 15:35.14 | voila | nope again |
| 15:35.41 | voila | http://pastebin.com/uhicShWD .. guys .. i try to call my softphone .. but didn't get any call |
| 15:35.50 | voila | i m new to this .. |
| 15:35.51 | voila | :( |
| 15:36.07 | aberrios_ | pastebin any Asterisk CLI output |
| 15:36.37 | voila | aberrios_ : is it for me ?? |
| 15:36.55 | aberrios_ | voila, yes, dial and pastebin the asterisk cli output when you call your softphone |
| 15:37.28 | voila | aberrios_: hey .. when i dial 1234 , then i m getting the incoming call |
| 15:38.06 | aberrios_ | voila, yes the default context for you softphone will include default1 so when you dial extension 1234 it executes Dial(SIP/3005) |
| 15:38.09 | voila | aberrios_: i mean ... when client call 1234 , then it get incoming call |
| 15:38.18 | voila | aberrios_: ok |
| 15:38.35 | voila | aberrios_: but i want something , so client get call without dialing anything |
| 15:38.48 | voila | i mean , u can say like , i want to make a call to client |
| 15:39.07 | aberrios_ | voila, and by client to mean softphone? |
| 15:39.12 | voila | yaa |
| 15:39.36 | voila | i want asterisk to call softphone , without client's dialling the extension number(1234) |
| 15:39.39 | aberrios_ | voila, so you want another phone to dial the softphone without them pressing any buttons? |
| 15:40.33 | voila | aberrios_: actaully i want this , when some event occurs on server side , then it informs to all clients (softphones) |
| 15:40.48 | aberrios_ | voila, right |
| 15:41.15 | aberrios_ | voila, then use call files. The event should trigger the creation of a call file which drops into a context say [ringall] |
| 15:41.23 | aberrios_ | voila, that context should contain something like: |
| 15:41.43 | voila | aberrios_: ? |
| 15:41.44 | aberrios_ | voila, exten => s,1,Dial(SIP/101&SIP/102&SIP/103( |
| 15:42.01 | *** join/#asterisk hfb (~hfb@pool-96-247-49-75.lsanca.dsl-w.verizon.net) |
| 15:42.10 | aberrios_ | voila, http://www.voip-info.org/wiki/view/Asterisk+Call+Files |
| 15:42.20 | voila | aberrios_ : ok .. got it .. i will gooogle it .. thanks |
| 15:42.47 | voila | one things more .. is it possible to send text mssge to softphone .. sryr if it wierd question |
| 15:43.12 | aberrios_ | voila, SMS? or like a Text CallerID? |
| 15:43.17 | voila | SMS |
| 15:43.25 | voila | the sms we send in real life |
| 15:43.42 | aberrios_ | voila, Which softphone supports SMS? |
| 15:43.57 | voila | aberrios_: nope , i m asking from u |
| 15:43.59 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:44.12 | voila | is it possible to send sms from asterisk to softphone ? |
| 15:44.25 | aberrios_ | voila, I don't think any softphone support SMS, but if its a softphone why not use a CallerID which contains a message |
| 15:44.41 | voila | ohh . ok .. fine |
| 15:44.54 | aberrios_ | voila, Asterisk can handle SMS but sending SMS to a softphone seems a bit redundant to me. |
| 15:44.56 | voila | and thanks a lot for ur suggestions advice and help |
| 15:45.01 | aberrios_ | voila, np |
| 15:45.04 | voila | okies . |
| 15:45.06 | voila | thanks |
| 15:45.11 | voila | takecare bye |
| 15:45.15 | aberrios_ | cya |
| 15:45.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:45.49 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:49.21 | aberrios_ | voila, You could also use AMI, but considering you're fairly new to asterisk I'd recommending starting out with call files. its easier. |
| 15:52.59 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 15:57.39 | *** join/#asterisk wonderworld (~ww@port-92-201-100-92.dynamic.qsc.de) |
| 16:02.51 | *** part/#asterisk voila (IceChat77@117.212.65.182) |
| 16:03.04 | *** join/#asterisk tstorm (~tstorm@173-164-230-21-SFBA.hfc.comcastbusiness.net) |
| 16:06.31 | m_tadeu | is there a place where I can copy the sql to create the CEL table in mysql? |
| 16:33.49 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:35.49 | aberrios_ | HOMETIME! wheeee, laters peep |
| 16:37.02 | breardo | hmm |
| 16:37.12 | breardo | i thought qualify time was dictated by qualifyfreq in sip.conf ? |
| 16:41.43 | *** join/#asterisk zkn (~zkn@c213-89-4-148.bredband.comhem.se) |
| 16:41.54 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 16:43.02 | atan | I am having some call issues using SIP on a mobile phone. Is there a way to tell what codecs asterisk has installed and able to use? |
| 16:46.54 | JerJer | atan: um, yeah |
| 16:48.19 | zkn | atan: core show translation or core show codecs audio could maybe help |
| 16:49.44 | atan | Ahh yes, core show codecs does produce a large list. |
| 16:49.59 | atan | I wonder which would be the best bet to use. |
| 16:50.08 | atan | bangs head on desk |
| 16:50.58 | atan | Right now it's set to 'ulaw,gsm' |
| 16:53.21 | *** join/#asterisk vutamhoan (~user@186.1.193.254) |
| 16:55.35 | vutamhoan | I'm using * with 2 interface - when Invite comes to 1st interface, * send 200 OK with 1st IP in sdp - but the RTP comes out with IP of 2nd interface - Please help |
| 17:00.28 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-135.dsl.net.pk) |
| 17:01.00 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 17:02.47 | mechbangirc | hi big brains. i have an asterisk box version 1.8.3, i setup odbc_adaptive for mysql to make use of CDR(custome) fields. everything is working fine but then i have these warnings http://pastebin.com/diS9DVjz , should i care about them |
| 17:04.01 | *** join/#asterisk rickcorp (~rickcorp@c-76-100-218-165.hsd1.md.comcast.net) |
| 17:06.10 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 17:11.40 | *** join/#asterisk IridiumScaffold (~Iridescen@124.169.155.140) |
| 17:12.18 | IridiumScaffold | does anyone know how to stop a voicetronix card from picking up outbound faxes and sending them to a fax extension? |
| 17:13.52 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:20.04 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:25.05 | anonymouz666 | anyone in here using master-master replication with Asterisk? |
| 17:25.14 | anonymouz666 | with mysql |
| 17:26.17 | JerJer | define with asterisk |
| 17:26.20 | JerJer | as in realtime |
| 17:26.40 | JerJer | i have done it with openser back in the day |
| 17:26.41 | anonymouz666 | NO |
| 17:27.00 | anonymouz666 | in fact, I juse want to know how these people make sure that the database is always sync each other. |
| 17:27.05 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:27.11 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 17:28.18 | JerJer | anonymouz666: for one, have an active monitoring situation place for WHEN mysql decides to corrupt your data or otherwise breaks replication |
| 17:30.13 | anonymouz666 | database is in sync the mysql1 dies the mysql2 is active, then we back the 1 and it is no more in sync with 2. |
| 17:30.37 | anonymouz666 | and that will force another downtime. |
| 17:30.45 | anonymouz666 | not smart |
| 17:30.51 | zkn | i am registering with the ITSP using "register => ...", CLI sip show registry states that the asterisk has registered with the account, however when I call the related DID, I get: Sending fake auth rejection for device ... ... |
| 17:30.57 | zkn | what could be the reason for that? |
| 17:31.51 | JerJer | mysql1 should check the replication logs (binlog?) and re-sync itself |
| 17:32.21 | anonymouz666 | that is done through binlog |
| 17:32.29 | anonymouz666 | but happens automatically when you setup the replication? |
| 17:33.08 | JerJer | in a perfect world |
| 17:33.13 | anonymouz666 | haha |
| 17:33.30 | vutamhoan | How can I force RTP is create on the same interface as sdp? |
| 17:33.31 | anonymouz666 | JerJer: in a perfect world my accounting would be done at sip proxy level. |
| 17:34.46 | JerJer | anonymouz666: i totally agree |
| 17:35.03 | JerJer | (and practice too) |
| 17:36.39 | vutamhoan | I got rtp problem, Invite comes to 1st interface, * send 200 OK with 1st IP in sdp - but the RTP comes out with IP of 2nd interface - Please help |
| 17:37.44 | anonymouz666 | vutamhoan: externip/localnet doesn't help in your case? |
| 17:38.00 | vutamhoan | No, we don't have NAT |
| 17:38.29 | vutamhoan | problem is we have 02 route for backup |
| 17:38.38 | anonymouz666 | externip change the SDP if localnet doesn't match. |
| 17:38.45 | anonymouz666 | well, don't know then |
| 17:40.02 | vutamhoan | For example, incomming INVITE may come to IP1 or IP2, * return 200 with correct sdp as receiving interface |
| 17:40.29 | vutamhoan | but RTP alway comes out through IP1 |
| 17:43.25 | *** join/#asterisk dev-ast (~dshah@utdpat242007.utdallas.edu) |
| 17:52.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 17:53.04 | dev-ast | In Asterisk 1.8.3, no matter what port number I define under peer configuration, UDP always sticks with default port 5060. I want my outgoing UDP on port port 5508. On applicaiton layer, it says that outgoing SIP conversation is on port 5508, but UDP says port 5060 |
| 17:57.59 | IridiumScaffold | is there no way to temporarily stop asterisk detecting faxes so I can send one outbound? |
| 17:58.37 | IridiumScaffold | I'm trying to use iaxmodem for send/receive. fax detection works fine and routes to the right extension, but when I try to dial outbound, it detects a fax and loops it back to the iaxmodem device |
| 17:59.25 | *** join/#asterisk eugeneoden (~goden@99-62-173-93.lightspeed.austtx.sbcglobal.net) |
| 18:03.58 | *** join/#asterisk frek818 (~herman@2001:470:f0f9:100:a6ba:dbff:feef:34b1) |
| 18:08.45 | Tozz_ | lol http://i.imgur.com/CVBHB.gif |
| 18:09.38 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:23.24 | *** join/#asterisk work2play_ (d141f2da@gateway/web/freenode/ip.209.65.242.218) |
| 18:23.41 | work2play_ | sorry not sure if that last one went through - chat locked up |
| 18:23.49 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:e0cf:c6ac:3af7:80ec) |
| 18:24.48 | work2play_ | but I wanted to know if anyone might be able to provide a little bit of guidance on the proper setup of dial patterns in my outbound routes vs number manipulation in the sip trunk propterties... I only have a single sip provider |
| 18:25.40 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 18:25.41 | work2play_ | basically I want to dial 9 for any outside lines as I normally would for a business pbx; and I want to auto-append my area code should I dial 7-digits since my sip provider needs it. |
| 18:33.21 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:35.29 | atan | Oh my god. I could ring a neck right now. Verizon must have a customer service phone number similar to mine and now I'm getting a ton of misdials... blarg! |
| 18:35.34 | cusco__ | work2play_: and what problems are you facing? |
| 18:36.13 | work2play_ | wasn't getting the local area code prepended when dial 9,NXXNXXXXXX |
| 18:36.13 | cusco__ | atan: add some IVR stating this is not verizon :P |
| 18:36.44 | atan | cusco__, always an option :D or redirect to Sprint |
| 18:36.49 | work2play_ | i think I may be catching on - testing some changes now |
| 18:36.54 | work2play_ | nice one ;-) |
| 18:37.35 | cusco__ | someone dials NXXNXXXXXX, dialplan has DIAL(000$EXTEN) |
| 18:37.44 | cusco__ | 000 being whatever |
| 18:39.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:ecab:5fb5:ffa0:2583) |
| 18:52.08 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 18:52.46 | *** join/#asterisk RickB17 (~rbreidens@pat.recoverynetworks.com) |
| 18:58.08 | frek818 | I have a pretty simple question about hour asterisk process SIP RTP. I have a server in the US and and sip phone the Philippines. When I call the phone the first 1/2 second of the audio is missing. However, tcpdump on the sip servers show that there is 0 packet loss and 0 out of sequence packets. The latency on the phone is 'sip show peers' is 332ms. |
| 18:59.26 | frek818 | Will asterisk ignore RTP from the phone until the final ACK is received for INVITE? |
| 19:00.15 | dev-ast | RTP will not come into play until final ACK is received for INVITE |
| 19:00.17 | frek818 | It would kind of make sense since the latency is about 400ms. |
| 19:01.09 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 19:01.58 | frek818 | dev-ast: Is that a definite answer? |
| 19:02.46 | dev-ast | yes.. as RTP session is not established until Ack is receieved |
| 19:03.46 | dev-ast | it's weird that you are missing 1/2 second of the audio |
| 19:04.15 | dev-ast | however 1/2 second is close to 400ms |
| 19:05.32 | frek818 | In my tcpdump, I can see that the softphone sends RTP before it send it's OK. |
| 19:05.37 | dev-ast | try to increase jitter buffer and see if you still have 1/2 second audio missing |
| 19:06.11 | dev-ast | hmmm..... |
| 19:07.13 | dev-ast | Do you have any SIP standard desk phones like Polycom 330s and 335s |
| 19:07.37 | frek818 | Yes, but not in the Phillipines. Just here in my office. |
| 19:07.42 | frek818 | in the US |
| 19:08.08 | dev-ast | you might want to run tcpdump for that and see if you can see RTP before OK |
| 19:08.29 | dev-ast | i believe it might be softphone not following SIP standard |
| 19:08.45 | frek818 | K, ill try that out. |
| 19:09.00 | frek818 | thx\ |
| 19:09.02 | dev-ast | ok.. |
| 19:09.07 | dev-ast | no problem |
| 19:18.21 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 19:18.23 | Godofmonkeys | I've got a quick question, my incoming plan reads "exten => 3038000142,n,Goto(myphones,100,1)" If I set a global to ring multiple lines, ${GRP1} would it be "GRP1=SIP/100&SIP/101&SIP/102&SIP/103&SIP104" or GRP1=100&101&102&103? |
| 19:18.42 | *** part/#asterisk IridiumScaffold (~Iridescen@124.169.155.140) |
| 19:18.54 | Godofmonkeys | then exten => 3038000142,n,Goto(myphones,${GRP1},1) |
| 19:19.04 | *** part/#asterisk e911_indiana (~medic@c-98-220-104-243.hsd1.in.comcast.net) |
| 19:20.45 | serafie | Godofmonkeys: the first one. Each group separated by "&" must contain the technology. |
| 19:21.24 | Godofmonkeys | serafie, then would exten => 3038000142,n,Goto(myphones,${GRP1},1) be correct? |
| 19:21.32 | serafie | no |
| 19:21.43 | serafie | You want to ring all those exxtensions at once? |
| 19:21.48 | Godofmonkeys | yes |
| 19:21.58 | serafie | You need to do Dial(${GRP1}) |
| 19:22.21 | frek818 | dev-ast: My hardphone doesn't send RTP before it receives the ACK from the server. |
| 19:22.26 | *** join/#asterisk ariel_ (~abatista@63.214.236.169) |
| 19:22.33 | Godofmonkeys | serafie, thanks |
| 19:22.39 | serafie | https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 19:22.59 | serafie | Goto() is for jumping to another extension. |
| 19:23.11 | serafie | *context/extension/priority |
| 19:26.44 | Godofmonkeys | exten => 3038000142,n,Dial(${GRP1},30) |
| 19:26.44 | Godofmonkeys | exten => 3038000142,n,VoiceMail(301,u) |
| 19:26.44 | Godofmonkeys | <PROTECTED> |
| 19:28.23 | *** join/#asterisk supjigator (~supjigato@mail.tektone.net) |
| 19:31.51 | serafie | Godofmonkeys: yep. I hope you've got a 1 priority, though. Can't start with "n." |
| 19:32.11 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 19:32.24 | Godofmonkeys | i do, it logs the incoming call |
| 19:32.36 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 19:34.37 | Godofmonkeys | so by this logic, i should keep the voicemail in each extension so when they are direct dialed the users will have thier voicemail |
| 19:39.02 | breardo | okay |
| 19:39.06 | breardo | WTF happens on a network EVERY 34 minutes? :) |
| 19:39.35 | breardo | my SIP peers are unqualifing with a spike up to 5500ms every 34 mins, then seconds later, re-qualify back to 12-15ms. |
| 19:39.37 | Freeaqingme_ | uhm, if you have a dhcp server with a bit of a weird leasetime |
| 19:39.44 | breardo | OMFG |
| 19:39.46 | breardo | LEASE TIME |
| 19:42.14 | Qwell | but it shouldn't spike unless many systems on the network hit the DHCP server for the first time, all simultaneously |
| 19:43.03 | Qwell | though I suppose the dhcpd could be really stupid on first lease, and give you a lease time of interval - time until interval |
| 19:43.13 | breardo | finally a reasonable response |
| 19:43.14 | breardo | thanks |
| 19:44.48 | breardo | :) |
| 19:45.11 | Qwell | breardo: If you find out what it is, I'd be curious to know. |
| 19:46.36 | breardo | i'll be sure to let you know |
| 19:47.14 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 19:47.20 | Qwell | strangely, that's 255*8 seconds |
| 19:47.47 | breardo | hmmm |
| 19:47.50 | Qwell | (or 256*8, if the 34 minutes is from rounding) |
| 19:47.55 | breardo | that is interesting |
| 19:53.56 | breardo | hmm |
| 19:54.11 | breardo | yeah its ~34 mins.. not 34 mins exactly |
| 19:55.02 | Qwell | get an epoch timestamp. I bet it (give or take 10ish) is divisible by 2048 |
| 19:56.16 | breardo | got em already |
| 19:56.17 | breardo | one sec |
| 19:59.58 | breardo | hmm.. i dunno |
| 20:00.26 | breardo | 1301583061 |
| 20:01.24 | breardo | fwiw, the DHCP leases are set up at 30 days on this site.. |
| 20:01.28 | breardo | 30 days.. grrr |
| 20:03.35 | Qwell | That's about 800s off. I wouldn't expect it to be anywhere near that high if my theory held. |
| 20:08.40 | breardo | well i coud be wrong, i accidentally had it store time parsed through date, not as unix time in tcpdump :| |
| 20:08.44 | breardo | so i reversed it |
| 20:08.45 | *** join/#asterisk Kottizen (kottizen@unaffiliated/icanhasfreenode) |
| 20:08.49 | *** part/#asterisk Kottizen (kottizen@unaffiliated/icanhasfreenode) |
| 20:09.01 | breardo | i'll catch a true epoch timestamp on the next iteration |
| 20:09.34 | *** join/#asterisk zooz (~zooz@host86-145-217-142.range86-145.btcentralplus.com) |
| 20:10.25 | breardo | also interesting.. I got an IP from a phone that's always affected, every 34 mins and ran a continuous ping against it, to correlate the upswing in latency that asterisk reports, with actual ping data; and lo and behold, the phone did not unqualify this time |
| 20:10.41 | breardo | cant say for certain its related after just one iteration, but i suspec this will hold true after several |
| 20:11.31 | breardo | also, fwiw, i upped my qualify timeout to 4s from 2s, and now it reports the phones as lagged first.. they all swing up to ~5200ms |
| 20:11.42 | zooz | I do have a situation here, my asterisk server is behind a NAT also the client is behind another NAT, I do have SIP and RTP data ports forwarding on the firewall where asterisk server resides, my SIP client uses a public STUN server, but I still cannot "hear" anything, what else needs to be done? |
| 20:12.29 | breardo | [1301601052.089242] 64 bytes from 10.3.2.44: icmp_req=861 ttl=64 time=1.31 ms |
| 20:12.31 | breardo | [1301601053.090585] 64 bytes from 10.3.2.44: icmp_req=862 ttl=64 time=1.31 ms |
| 20:12.34 | breardo | [1301601097.092603] 64 bytes from 10.3.2.44: icmp_req=906 ttl=64 time=1.82 ms |
| 20:12.35 | breardo | DOH |
| 20:12.37 | breardo | sorry |
| 20:12.40 | breardo | check out the packet loss.. |
| 20:12.46 | breardo | doesnt swing up.. just dies for ~30 packets |
| 20:12.51 | breardo | err ~40 |
| 20:13.18 | seraphie | breardo: hail mary, here, but did you check the crontab? |
| 20:13.51 | breardo | well.. no.. good call |
| 20:13.53 | breardo | i'll check it |
| 20:14.09 | seraphie | worth a shot. :-\ |
| 20:14.26 | breardo | aye... nah theres nothing |
| 20:14.33 | breardo | just the hourly / daily reporting.. |
| 20:15.33 | *** part/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 20:17.18 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
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| 20:21.38 | p3nguin | seraphie: You don't ring extensions, you ring devices. The extension contains the Dial command to make it happen. |
| 20:22.49 | seraphie | p3nguin: I am certain that I didn't say that, because I know better. ;) It was that dumb serafie. |
| 20:24.35 | zooz | I do have a situation here, my asterisk server is behind a NAT also the client is behind another NAT, I do have SIP and RTP data ports forwarding on the firewall where asterisk server resides, my SIP client uses a public STUN server, but I still cannot "hear" anything, what else needs to be done? |
| 20:26.40 | breardo | man this is a head scratcher |
| 20:26.44 | breardo | asking in #cisco now too :) |
| 20:27.18 | *** join/#asterisk guntbert (~re@unaffiliated/guntbert) |
| 20:27.25 | *** part/#asterisk guntbert (~re@unaffiliated/guntbert) |
| 20:30.04 | Qwell | oh, Cisco switches? |
| 20:30.06 | Qwell | there's your problem |
| 20:34.31 | zooz | who said cisco switches? |
| 20:36.34 | *** join/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 20:36.40 | breardo | haha |
| 20:36.47 | epart | hi guys |
| 20:37.26 | epart | im a nwbe.. i heard about asterisk .. open source pabx.. |
| 20:37.53 | epart | kindly help me on how to setup it in ubuntu 10.04.1 |
| 20:38.19 | Qwell | ~newbook |
| 20:38.19 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 20:38.23 | Qwell | epart: There you go. |
| 20:39.45 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 20:39.49 | epart | thnk you |
| 20:42.54 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 20:45.12 | *** join/#asterisk slidesinger-lt (~jtatum@12.149.62.10) |
| 20:45.12 | breardo | the cisco guys pointed out that 34m is nearly 2000 seconds |
| 20:50.28 | *** part/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 20:54.05 | zooz | if anyone is wondering how I got both asterisk behind NAT and clients behind another NAT working - give me a shout |
| 20:56.28 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 20:57.40 | p3nguin | You can share your story, but it probably doesn't differ much from anyone else with similar physical configuration. |
| 20:59.10 | zooz | p3nguin, why no one answered my question then when I asked about the same issue? |
| 20:59.30 | p3nguin | I wouldn't have any idea, because I wasn't here. |
| 20:59.43 | zooz | right |
| 20:59.52 | p3nguin | If you had special needs and required a special software configuration, post it on the wiki. |
| 21:00.21 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 21:00.51 | zooz | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:01.07 | zooz | above does not contain the solution I found to be working |
| 21:01.13 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 21:01.42 | p3nguin | Under normal circumstances, you would configure Asterisk to work behind NAT, then configure each peer according to its NAT requirement. Forward to Asterisk UDP port 5060 and the UDP port range defined in rtp.conf. Do not forward anything on the phone's network and do not configure the phone for any NAT traversal. |
| 21:02.02 | p3nguin | ^ That is a typical configuration for Asterisk behind NAT and phones behind another NAT. |
| 21:02.08 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:03.30 | p3nguin | Some people can't keep their hands off the phone's NAT settings, and that's where they go wrong. If you configure the phone's NAT settings, it breaks Asterisk's ability to NAT the peer. |
| 21:04.12 | *** join/#asterisk kaelef (~kaelef@174-18-235-23.tcso.qwest.net) |
| 21:04.32 | *** join/#asterisk cashback (~cashback@ip68-2-140-46.ph.ph.cox.net) |
| 21:04.46 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 21:10.41 | p3nguin | Other people get the bright idea to play with port forwarding on the phone's network. That usually doesn't turn out too good either. |
| 21:11.20 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 21:11.29 | p3nguin | Then there are the morons who think DMZ is the magical solution for phones and/or Asterisk. |
| 21:13.48 | *** join/#asterisk citrusfizz (~chatzilla@sb1.serviceobjects.com) |
| 21:14.14 | citrusfizz | for an office pbx whats better. elastix or asteriskNOW/freepbx |
| 21:16.15 | p3nguin | citrusfizz: If you have only elastix and AsteriskNOW to choose from, AsteriskNOW is much better. |
| 21:18.27 | p3nguin | citrusfizz: But if you also have plain Asterisk on your favorite Linux OS, that's my preference. |
| 21:18.53 | p3nguin | s/OS/OS to choose from/ |
| 21:30.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:31.33 | *** join/#asterisk godofmonkeys (~Godofmonk@71-87-181-173.dhcp.jcsn.tn.charter.com) |
| 21:33.35 | *** join/#asterisk ccesario (~ccesario@201-95-147-176.dsl.telesp.net.br) |
| 21:35.22 | godofmonkeys | my callerid is hosed; shows the proper name but the number is the username of my account with my provider. somebody point me to the proper place? |
| 21:35.23 | *** join/#asterisk [loy] (~nobody@95.72.54.156) |
| 21:37.09 | godofmonkeys | incoming* |
| 21:40.17 | citrusfizz | p3nguin: i am currently using trixbox CE, i couldn't imagin using all the features we use and having the same functionality with just plain asterisk, that would drive me insane to setup |
| 21:44.50 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 21:54.40 | *** join/#asterisk JonnyD_work (~Jon@64.135.99.32) |
| 21:58.19 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.191.92.93) |
| 22:00.18 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 22:10.32 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 22:15.24 | p3nguin | godofmonkeys: Are you talking about inbound or outbound calls? |
| 22:16.34 | Greek-Boy | What would you guys choose between Trixbox and Elastix? And why... ? |
| 22:18.17 | p3nguin | If I only had those two choices? |
| 22:19.15 | p3nguin | If I only had those two choices, I'd probably shoot myself. |
| 22:19.18 | _Corey_ | hahaha |
| 22:19.19 | Greek-Boy | lol |
| 22:19.19 | Greek-Boy | ok |
| 22:19.23 | p3nguin | Neither one is a good choice. |
| 22:19.27 | Greek-Boy | so what would be your first choice then? |
| 22:19.31 | p3nguin | Asterisk. |
| 22:19.32 | Greek-Boy | plain Asterisk? |
| 22:19.39 | p3nguin | On my favorite distro. |
| 22:20.00 | p3nguin | But if you MUST go with a ready-to-run solution with FreePBX, AsteriskNOW is the correct choice. |
| 22:20.30 | Greek-Boy | the thing is this... |
| 22:20.44 | Greek-Boy | my admins aren't into linux and manual confing |
| 22:20.47 | Greek-Boy | unfortunately... |
| 22:21.07 | Greek-Boy | so that is why they have suggested that I let them run elastix |
| 22:21.29 | p3nguin | (1719.59) <p3nguin> But if you MUST go with a ready-to-run solution with FreePBX, AsteriskNOW is the correct choice. |
| 22:21.39 | p3nguin | See above. |
| 22:22.01 | p3nguin | Elastix is junk and you won't get any support for it. |
| 22:22.39 | Greek-Boy | ok |
| 22:23.00 | p3nguin | AsteriskNOW isn't junk, this channel can support the Asterisk part of it and the FreePBX channel can support the FreePBX part of it, and the AsteriskNOW channel can support it in its entirety. |
| 22:24.07 | Greek-Boy | i see |
| 22:24.23 | Greek-Boy | but just because something doesn't have good support it doesn't mean it's junk right? |
| 22:24.30 | p3nguin | no |
| 22:24.43 | p3nguin | But being junk does make it junk. |
| 22:25.00 | Greek-Boy | lol |
| 22:25.00 | Greek-Boy | ok |
| 22:25.05 | p3nguin | And not having support means you won't get support for the junk. |
| 22:25.53 | Greek-Boy | ok |
| 22:25.54 | Greek-Boy | thanks |
| 22:25.54 | Greek-Boy | :-) |
| 22:26.13 | *** join/#asterisk brainiac (~brainiac@208.86.215.38) |
| 22:29.25 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 22:32.31 | Sertys | what's the best free billing? |
| 22:32.36 | Sertys | for asterisk |
| 22:39.52 | *** join/#asterisk VoIPBroker (~VoIPBroke@96.176.19.11) |
| 22:44.42 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
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| 23:08.02 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 23:08.55 | VoIPBroker | Hey everyone. Hope everything's going good with those boxes.. If you need some US termination, let me know. Here's our rates http://tinyurl.com/6y63shv if you want to test quality, send me a private message |
| 23:12.28 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 23:17.01 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 23:30.39 | p3nguin | voipbroker: Too bad there's no sign-up on the site. I wouldn't mind going for a test drive, but without the ability to sign myself up, I quickly lose interest. |
| 23:31.26 | VoIPBroker | I'm your sign in, private message me. |
| 23:32.19 | p3nguin | Like I said, without the ability to sign myself up on the site, I quickly lose interest. |
| 23:34.33 | VoIPBroker | Have fun getting rape at Gafachi. |
| 23:35.18 | p3nguin | *shrug* no idea |
| 23:36.47 | therawr | irc is CLEARLY the best way to broker clientelle |
| 23:37.49 | p3nguin | If I could simply sign up on the web site, by myself, when I feel like doing it, I would go for a test drive. |
| 23:38.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:38.10 | p3nguin | And if I liked the service, away we'd go. |
| 23:38.19 | iprouteth0 | with google voice integration in 1.8, I no longer even need a provider for my setup, but my use case differs i'm sure from a business scenario |
| 23:38.39 | p3nguin | But there's no sign up, so I probably won't ever know if the service is any good. |
| 23:39.08 | p3nguin | I have to assume, based on the customer service I've received, that it sucks. |
| 23:39.31 | therawr | p3nguin +1 |
| 23:40.02 | p3nguin | Well, you know, when people have shit to work with, it often makes them grumpy. |
| 23:40.20 | iprouteth0 | in VoIP, I really agree with you p3nguin. being able to sign up without contacting anyone on the phone to do so and get up and running is par for the course in ITSP |
| 23:40.54 | p3nguin | Internet services everywhere are that way, not just VoIP services. |
| 23:41.21 | p3nguin | Sign up on the site, get an email, activate, etc. |
| 23:41.35 | iprouteth0 | true, though I've definitely noticed the trend is strong in voip |
| 23:42.25 | p3nguin | When was the last time you had to make a phone call or PM someone on IRC to sign up for some random forums or bulletin boards? |
| 23:42.33 | tzanger | !itsp-us |
| 23:42.42 | tzanger | ?itsp-us |
| 23:42.48 | tzanger | how do you invoke the bot, I can't remember |
| 23:42.58 | p3nguin | ~itsplist-us |
| 23:42.58 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:43.04 | iprouteth0 | there is still an overwhelming amount of customers who would much rather speak to someone on the phone... |
| 23:43.26 | p3nguin | And for that, there's typically a customer service phone number on the site. |
| 23:44.20 | iprouteth0 | oh indeed. It's what I do. I'm in all queues for our support :( |
| 23:44.44 | *** join/#asterisk QbY (~kelvin@96.176.19.11) |
| 23:44.48 | *** join/#asterisk frek818 (~herman@2001:470:f0f9:100:a6ba:dbff:feef:34b1) |
| 23:45.34 | iprouteth0 | we serve areas of many of the southern US states though so I guess it stand to reason that lots of people prefer 'legacy' support |
| 23:46.35 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 23:46.45 | p3nguin | That site doesn't have a sign up NOR a customer service number. To me, that says they aren't interested in doing business. |
| 23:47.17 | iprouteth0 | lack of tel# is massively bad |
| 23:47.46 | QbY | p3nguin: ever worked for a manager who jumps the gun? |
| 23:47.56 | iprouteth0 | how often do people come in here to try to sell stuff? |
| 23:49.00 | QbY | iprouteth0: have you ever worked for a manger who jumps the gun? |
| 23:49.19 | p3nguin | It happens from time to time. |
| 23:50.04 | iprouteth0 | QbY: well, in some ways our company sometimes does and my manager has to sometimes has to come up with a workaround due to that |
| 23:50.19 | QbY | ok, so here's the deal.. i work in the same office with sean. my job is to get 855-767-2577 operational--a new number for acsc, which is a division of acgi.. then once its done, i'm to tell my co-worker is ready then he's to activate all the contact shit. |
| 23:50.20 | iprouteth0 | but not for a specific manager that did so |
| 23:50.52 | iprouteth0 | ah, the sensitive topic of cutovers |
| 23:50.52 | QbY | however, i haven't gotten it done, and some fucking prick with more intelligence than we can measure--becase he's management--has called over to sean's desk bitching that he HAS to get 10 trial activated today |
| 23:51.10 | iprouteth0 | is this occurring in a maintenance window? |
| 23:51.26 | QbY | so poor little sean, with no experience with pricks on irc, has jumped on after asking me to help--thereby delaying the number activation--came in here to hand out trials |
| 23:51.41 | QbY | he's trying to save his job, by giving away trials.. |
| 23:51.44 | QbY | he's not asking for your money |
| 23:51.49 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 23:51.56 | QbY | he's trying to give you a few hundred/thousand minutes (depdning on where you call) |
| 23:52.14 | QbY | so, go blast some mailing list, and leave the dude alone |
| 23:52.15 | iprouteth0 | I'm sorry to hear about your predicament |
| 23:52.18 | QbY | thanks |
| 23:52.21 | *** join/#asterisk ariel_ (~abatista@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 23:52.26 | ariel_ | hello folks |
