| 00:04.23 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
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| 00:56.11 | ducdmann | hi powerunits,what u need ? |
| 00:56.42 | *** join/#asterisk coppice (~coppice@123203240234.ctinets.com) |
| 01:30.18 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
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| 01:45.44 | *** join/#asterisk gunnarar (~gunnarar@96.53.60.126) |
| 01:46.23 | gunnarar | I upgraded my asterisk box from 1.6 to 1.8 and now most of my cisco 7970 phones don't work |
| 01:46.49 | gunnarar | I'm running firmware 8.5.2 and 8.2.4 on them |
| 01:47.30 | gunnarar | I get 401 unauthorized but I'm positive that I'm using the correct user and pass and the settings on the phones have not been changed in the meanwhile |
| 01:47.41 | gunnarar | can someone here help me with that? |
| 01:48.03 | *** join/#asterisk RickCogley (~anonymous@EM1-113-238-95.pool.e-mobile.ne.jp) |
| 01:48.27 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-136-96.ks.ks.cox.net) |
| 01:51.50 | gunnarar | is there perhaps another channel I should be going to for this? |
| 02:15.04 | ducdmann | check your extensions.conf |
| 02:15.24 | ducdmann | seems something is missing there |
| 02:16.01 | ducdmann | I did an upgrade and had similar issues |
| 02:16.36 | gunnarar | can you recall what you found there? |
| 02:16.51 | *** join/#asterisk RickCogley (~anonymous@EM1-113-238-95.pool.e-mobile.ne.jp) |
| 02:17.30 | ducdmann | most of the extensions were wiped out |
| 02:17.49 | ducdmann | apart from default asterisk exts |
| 02:19.22 | gunnarar | but I can log in from a softphone as that user... |
| 02:20.26 | ducdmann | some settings r missing after the upgrade |
| 02:20.46 | ducdmann | so u check sip.conf too |
| 02:21.06 | gunnarar | ok, I did back them up so diff I guess :) |
| 02:21.17 | gunnarar | thanks, I'll be back in a bit! |
| 02:21.39 | *** join/#asterisk scgm11 (~Sebastian@r186-52-44-122.dialup.adsl.anteldata.net.uy) |
| 02:23.48 | *** part/#asterisk scgm11 (~Sebastian@r186-52-44-122.dialup.adsl.anteldata.net.uy) |
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| 02:25.07 | *** part/#asterisk scgm11 (~Sebastian@r186-52-44-122.dialup.adsl.anteldata.net.uy) |
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| 02:28.45 | gunnarar | sip and extensions.conf are identical to the ones I have backed up |
| 02:30.00 | gunnarar | I am running a freepbx but I don't think that is a factor here |
| 02:30.26 | gunnarar | REGISTER sip:192.168.2.176 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.106:5060;branch=z9hG4bKa50bd3e1 From: <sip:129@192.168.2.176>;tag=001aa264fa91000266557d5c-455391bb To: <sip:129@192.168.2.176> Call-ID: 001aa264-fa910002-bda4965e-4acb4b85@192.168.2.106 Max-Forwards: 70 Date: Tue, 05 May 2009 20:40:18 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7971G-GE/8.5.3 Contact: <sip:129@192.168.2.106:5060; |
| 02:30.27 | gunnarar | transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001aa264fa91>";+u.sip!model.ccm.cisco.com="119" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:0 Name=SEP001AA264FA91 Load=term71.default Last=" Expires: 3600 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.106:5060;branch=z9hG4bKa50bd3e1;received=192.168.2.106;rport=49156 From: |
| 02:30.27 | gunnarar | <sip:129@192.168.2.176>;tag=001aa264fa91000266557d5c-455391bb To: <sip:129@192.168.2.176>;tag=as51b5d498 Call-ID: 001aa264-fa910002-bda4965e-4acb4b85@192.168.2.106 CSeq: 101 REGISTER Server: FPBX-2.10.0rc1(1.8.9.2) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="358472c8" |
| 02:30.27 | gunnarar | Content-Length: 0 |
| 02:31.00 | gunnarar | this is a wireshark sniff |
| 02:31.10 | gunnarar | it goes on like this forever |
| 02:31.48 | gunnarar | at the same time I can log in as that user using softphone |
| 02:36.23 | *** part/#asterisk scgm11 (~Sebastian@r186-52-44-122.dialup.adsl.anteldata.net.uy) |
| 02:46.06 | ducdmann | the phones r registered on asterisk as extensions? |
| 02:47.21 | ducdmann | u can run sip show peers in freepbx cli or asterisk cli |
| 02:47.56 | ducdmann | then u can see which phones r registered |
| 02:48.19 | gunnarar | yes, it doesn't register |
| 02:48.32 | ducdmann | cos it seems the username or passwords r corrupt |
| 02:49.12 | ducdmann | take one phone and enter new password-both on the phone and on freepbx |
| 02:49.35 | ducdmann | and see if it registers |
| 02:51.28 | ShaunR | is there a way to create entrys in the queue_log? |
| 02:53.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 03:14.54 | *** join/#asterisk K-Man`` (~dcr@syscom.lnk.telstra.net) |
| 03:15.29 | K-Man`` | hi all, im wondering if someone can help me with an "unreachable' issue with Cisco 7970 IP phones |
| 03:21.38 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 03:25.33 | leifmadsen | ~nat |
| 03:25.33 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 03:26.06 | leifmadsen | infobot: tell ducdmann about pb |
| 03:26.23 | K-Man`` | hi leifmadsen...that is setup properly |
| 03:26.47 | K-Man`` | the issues is that the phones worked yesterday, then 1 by 1 started getting the "unreachable" status and all dropped off |
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| 03:29.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:32.41 | gunnarar | <ducdmann> I created a new extension, I get the same response |
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| 03:44.49 | ducdmann | hmm......... |
| 03:45.17 | ducdmann | can u ping the ip of your freepbx server? |
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| 04:02.08 | *** join/#asterisk mbeierl (~mark@184.151.115.95) |
| 04:04.02 | mbeierl | not an asterisk problem, but maybe someone has some ideas: I use Vonage for VoIP services, and have changed ISPs to Bell Canada's 3g service. While call quality is good, I have an RTP setup problem. Usually the call will ring through, but then I get no audio in either direction. Wireshark shows the RTP packets going out, but nothing flowing back. I am forced to use a Netgear MBR1210, which, I think, is the problem. |
| 04:04.34 | mbeierl | I think it's got a broken SIP ALG port or something...? I just don't know how to even being troubleshooting this. and Bell is no help at all. |
| 04:08.46 | p3nguin | Always disable ALG. |
| 04:09.01 | p3nguin | Asterisk is not compatible with ALG. |
| 04:10.09 | [TK]D-Fender | And it isn't an asterisk problem |
| 04:10.16 | [TK]D-Fender | and this isn't #vonage |
| 04:10.16 | *** join/#asterisk K-Man`` (~dcr@syscom.lnk.telstra.net) |
| 04:10.48 | mbeierl | [TK]D-Fender: I know it's not. I was just asking to see if anyone had troubleshooting help. thanks anyways |
| 04:11.24 | K-Man`` | can someone please help? i have cisco 7970 ip phone with an "unreachable" status...while soft phones are working fine... i understand this maybe NAT related, but i think i have the nat setup ok. any help is appreciated |
| 04:13.25 | *** part/#asterisk mbeierl (~mark@184.151.115.95) |
| 04:14.23 | [TK]D-Fender | Unreachable means it failed to answer |
| 04:14.39 | K-Man`` | <[TK]D-Fender>, the phones register ok though |
| 04:14.50 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 04:15.13 | [TK]D-Fender | doesn't matter |
| 04:15.29 | [TK]D-Fender | means it didn't answer a qualify or didn't respond to a call or the reg timed out |
| 04:16.32 | K-Man`` | ok, i have restricted the port on the IP phone and set a static ip address...portforwarded that port to taht IP address... i also portforwarded that same port on asterisk side's router |
| 04:16.53 | K-Man`` | this worked last week..then yesterday all the phones started getting unreachable |
| 04:17.11 | K-Man`` | am i missing something? |
| 04:18.08 | [TK]D-Fender | None of that proves anything. |
| 04:19.11 | [TK]D-Fender | And you shouldn't be port forwarding on the phone side |
| 04:19.36 | K-Man`` | apparently that was the only way to get those 7970's to work..and they did for a week :) |
| 04:20.15 | *** join/#asterisk nzkiwi1 (~michaelnz@ip210-48-102-29.nettrust.net.nz) |
| 04:21.08 | K-Man`` | [TK]D-Fender: what should i be looking for? |
| 04:21.41 | p3nguin | If they worked for a week, what could possibly make them not work today? |
| 04:22.11 | K-Man`` | thats whats making me pull my hair out...nothing changed over the weekend and on monday they started dropping off one by one |
| 04:22.26 | [TK]D-Fender | K-Man``: Same things as always. SIP debug |
| 04:22.34 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 04:22.35 | nzkiwi1 | hi. can someone here recommend a US DID and VoIP provider? I want a good and competent one |
| 04:23.27 | p3nguin | nzkiwi1: Flowroute and VoIP.ms |
| 04:23.45 | p3nguin | I use VoIP.ms primarily. |
| 04:24.10 | nzkiwi1 | thanks P3 |
| 04:24.48 | nzkiwi1 | can these companies do the normal w/s stuff like CLI and trunking? |
| 04:24.58 | nzkiwi1 | (sip peering) |
| 04:25.32 | K-Man`` | [TK]D-Fender: are you able to look at the debug info that i collected? im not sure i can make alot of sense out of it |
| 04:25.46 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:26.14 | [TK]D-Fender | I can't look. You haven't shown any of it yet |
| 04:26.35 | K-Man`` | lol i mean if i make it available |
| 04:28.32 | K-Man`` | [TK]D-Fender: http://pastebin.com/7wdZW0yN - thanks in advance |
| 04:28.54 | *** join/#asterisk GameGamer43 (u5533@gateway/web/irccloud.com/x-jfadaovrilzwkthq) |
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| 04:30.03 | nzkiwi1 | voip.ms looks like a company I will investigate |
| 04:32.06 | [TK]D-Fender | K-Man``: Reliably Transmitting (no NAT) to 192.168.168.204:10004: |
| 04:32.15 | [TK]D-Fender | K-Man``: Contact: <sip:asterisk@202.167.246.57:5060> |
| 04:32.34 | [TK]D-Fender | K-Man``: You phone is on a LOCAL subnet yet you are giving them the PUBLIC address to return on |
| 04:32.41 | [TK]D-Fender | K-Man``: Your NAT setups is very wrong |
| 04:33.59 | *** join/#asterisk gunnarar (~gunnarar@96.53.60.126) |
| 04:34.01 | K-Man`` | [TK]D-Fender: Can i force the contact address? |
| 04:34.12 | [TK]D-Fender | K-Man``: You messed up your localnets |
| 04:34.19 | [TK]D-Fender | fix them |
| 04:34.25 | K-Man`` | in the sip.conf? |
| 04:34.33 | [TK]D-Fender | ... |
| 04:34.53 | [TK]D-Fender | yes |
| 04:35.22 | K-Man`` | in the sip.conf i have "localnet = <subnet of the asterisk server/subnet mask>" |
| 04:35.27 | K-Man`` | do i need to add any more? |
| 04:37.12 | [TK]D-Fender | K-Man``: you aren't showing actual configs. You also said "I think everything is right. |
| 04:37.22 | [TK]D-Fender | K-Man``: this isn't getting you anywhere it seems |
| 04:37.35 | K-Man`` | im just trying to understand.. |
| 04:37.52 | [TK]D-Fender | K-Man``: You aren't showing me anything real and you're asking me if it is RIGHT |
| 04:37.53 | K-Man`` | this is what i have... localnet = 10.200.24.0/255.255.255.0 |
| 04:37.58 | [TK]D-Fender | How do I know you did it right? |
| 04:39.08 | K-Man`` | just to clarify... 202.167.246.57 is the external ip address of the asterisk server |
| 04:39.23 | [TK]D-Fender | K-Man``: and where is 192.168.168.204:10004: <----------------- |
| 04:39.33 | [TK]D-Fender | Sure looks like a PRIVATE subnet you did not define |
| 04:39.55 | K-Man`` | 192.168.168.0/24 is the office subnet |
| 04:39.56 | [TK]D-Fender | What acounts for him? |
| 04:40.15 | [TK]D-Fender | Yes and why don't you have a LOCANET to say it is LOCAL? |
| 04:40.15 | K-Man`` | 10.200.24.0/24 is a remote office where asterisk is sitting |
| 04:40.46 | K-Man`` | i suspected i had to , but i wasnt sure |
| 04:41.16 | [TK]D-Fender | You do |
| 04:41.43 | [TK]D-Fender | If htey contact your subnet directly and you can return as such then they are local and you have to define them. all of them |
| 04:43.20 | K-Man`` | cool, thanks for that |
| 04:43.43 | K-Man`` | now i have ... |
| 04:43.44 | K-Man`` | localnet = 10.200.24.0/255.255.255.0 |
| 04:43.44 | K-Man`` | localnet = 192.168.168.0/255.255.255.0 |
| 04:43.57 | K-Man`` | but still unreachable |
| 04:45.54 | [TK]D-Fender | K-Man``: Just changing that doesn't make the phone re-register and keep in touch |
| 04:46.13 | [TK]D-Fender | the last theing the phone was told was the wrong place to call back |
| 04:46.16 | K-Man`` | i unregistered and registered again |
| 04:47.27 | K-Man`` | not enough? |
| 04:51.35 | K-Man`` | i even unregistered, restarted the phone :) |
| 04:54.00 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:00.33 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wqpoaglrsitlspfp) |
| 05:10.07 | p3nguin | nzkiwi1: To my knowledge Flowroute does not do trunking, but VoIP.ms does support IAX2 with trunking. |
| 05:11.40 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:12.12 | *** part/#asterisk shelest (~shelest@cpe-24-193-120-25.nyc.res.rr.com) |
| 05:12.14 | p3nguin | k-man``: If 10.200.24.0/255.255.255.0 is not local, do not define it in localnet. |
| 05:12.44 | K-Man`` | 10.200.24.0 is the subnet where the asterisk server is sitting |
| 05:13.01 | K-Man`` | so do i include it or not? |
| 05:13.05 | p3nguin | Okay, then if 192.168.168.0/255.255.255.0 is not local, do not define it in localnet. |
| 05:13.15 | p3nguin | You don't define localnets that are local to asterisk. |
| 05:13.36 | K-Man`` | i've taken it out |
| 05:13.39 | p3nguin | You don't need to define all RFC1918 nets that are in use in other places. |
| 05:14.50 | K-Man`` | i thought thats right |
| 05:16.13 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 05:16.15 | K-Man`` | im back where i started now though |
| 05:16.32 | p3nguin | Wait, what did I just type? |
| 05:16.34 | K-Man`` | where in the qualify process are the phones going wrong? |
| 05:16.43 | p3nguin | You don't define localnets that areN'T local to asterisk. |
| 05:16.55 | p3nguin | Not sure how I screwed up that. |
| 05:16.57 | K-Man`` | :) thats what i understood from you |
| 05:19.17 | p3nguin | I think I was going to say "don't define localnets that aren't local to asterisk," and ended up changing thoughts to "only define localnets that are local to asterisk" mid-sentence. |
| 05:19.35 | p3nguin | I hate when that happens. |
| 05:19.38 | K-Man`` | dont beat urself about it too much mate..its all good :) |
| 05:20.02 | K-Man`` | can you clarify something for me though.. |
| 05:20.26 | K-Man`` | " Contact: <sip:asterisk@202.167.246.57:5060> " <-- is this the address that the phones will reply on? |
| 05:20.28 | p3nguin | Did you read any articles about your phone +asterisk +nat? |
| 05:20.55 | p3nguin | Is asterisk a gateway with that address? |
| 05:20.58 | K-Man`` | i read plenty of articles and evntually got them registering, making and receiving calls...then yesterday they stopped |
| 05:21.18 | K-Man`` | asterisk is a server sitting behind a FW with that address...with 5060 forwarded |
| 05:21.36 | p3nguin | And what I meant by asterisk being a gateway was if the gateway is the one running asterisk. |
| 05:21.58 | p3nguin | So asterisk only has a private address. |
| 05:22.15 | K-Man`` | i did define the public ip address inthe sip.conf |
| 05:22.32 | K-Man`` | externaddr = 202.167.246.57:5060 |
| 05:24.01 | K-Man`` | to answer your question specifically...yes...asterisk has a private address |
| 05:24.33 | p3nguin | I'd have to see the entire call to figure out what's going on, and, quite frankly, I'm so tired I doubt I could decipher a sip debug right now. |
| 05:25.31 | p3nguin | I can't remember when I expect to see the public address and when I expent to see the private address in a Contact line. |
| 05:25.53 | K-Man`` | not to worry mate... i appreciate your help... get some rest |
| 05:26.06 | p3nguin | I intend to do that shortly. |
| 05:26.38 | K-Man`` | i'll keep pluging away at this one |
| 05:27.28 | p3nguin | I don't have experience with the 7970 with SIP and NAT, but I know it gives everyone problems. |
| 05:27.45 | K-Man`` | can you put that in writing for my boss? lol |
| 05:27.48 | p3nguin | The problem is always related to the source port. |
| 05:28.27 | K-Man`` | i forced the voipcontrolport on the phone using the config file and forwarded that specific port to the static ip address of the phone |
| 05:28.31 | *** join/#asterisk luke0512 (~eric@HSI-KBW-091-089-189-101.hsi2.kabel-badenwuerttemberg.de) |
| 05:28.34 | K-Man`` | this is whats getting it to register |
| 05:28.36 | p3nguin | The 7970 expects to get data from a specific port or something like that. It's like it expects the source port to be 5060, but in reality, the source port is fairly random. |
| 05:28.59 | p3nguin | You should never have to forward ports to a phone. Ever. |
| 05:29.11 | K-Man`` | there was no other way to get it to register |
| 05:29.16 | K-Man`` | what are the alternatives? |
| 05:29.30 | p3nguin | Asterisk's nat configuration and a NAT that isn't broken. |
| 05:29.56 | p3nguin | Are you using some crappy plastic router? |
| 05:31.32 | p3nguin | I'm trying to recall what the deal is with the ports... |
| 05:31.54 | K-Man`` | hahaha its a cisco 857 |
| 05:32.00 | p3nguin | I'm thinking the phone expects to receive the data back onto its own source port. |
| 05:32.27 | p3nguin | So if the phone sent from 16321, it expects the data back on that one instead of 5060. |
| 05:32.31 | p3nguin | Something weird like that. |
| 05:32.54 | p3nguin | Is any of this sounding familiar yet? |
| 05:32.55 | K-Man`` | is this from a range of media ports? |
| 05:33.22 | p3nguin | If I remember right, it's the SIP stuff, not the RTP stuff. |
| 05:33.36 | p3nguin | I was just using that port as an example. |
| 05:33.40 | K-Man`` | all sip ports are set to be 5060 |
| 05:34.02 | K-Man`` | the only port that edited was the VOIP control port |
| 05:34.03 | luke0512 | good morning |
| 05:34.06 | K-Man`` | which i restricted |
| 05:34.12 | K-Man`` | which i restricted manually |
| 05:34.15 | p3nguin | I've never once seen anyone come here with a 7970 working. |
| 05:34.58 | K-Man`` | did anyone get them to work for a week and then stop? haha |
| 05:35.08 | p3nguin | That doesn't sound familiar. |
| 05:36.13 | p3nguin | I had a Cisco 8xx something. I had to sell it. |
| 05:36.24 | p3nguin | It would not cooperate with Asterisk and RTP. |
| 05:36.41 | p3nguin | I don't think it was an 850 series. |
| 05:36.43 | K-Man`` | do you recommend a router that may help? |
| 05:37.17 | p3nguin | Can you build your own? Vyatta is pretty nice and is Asterisk compatible. |
| 05:37.40 | p3nguin | And Linux-based firewall/router should work. |
| 05:38.38 | K-Man`` | whats really confusing me is the fact that they worked for a little while.... it just doesnt make sense |
| 05:38.59 | K-Man`` | i'll explore building the linux fw/router, but im not sure if management will go for it |
| 05:39.02 | p3nguin | I have three different firewall routers that I use: one is RHL-based, one is CentOS (RHEL-based), and one is Vyatta (Debian-based). They all work great with Asterisks and phones behind NATs. |
| 05:39.34 | p3nguin | But you've still got the unfriendly 7970. |
| 05:41.16 | p3nguin | Okay, the wife says it's my bedtime. |
| 05:41.22 | K-Man`` | hmmmm |
| 05:41.35 | K-Man`` | better listen to the wife |
| 05:41.43 | K-Man`` | :) |
| 05:41.54 | p3nguin | Look for info on that port problem with the 7970. Perhaps that can give you some insight. |
| 05:42.24 | K-Man`` | cheers...ill keep on searching |
| 05:42.29 | K-Man`` | i appreciate your help |
| 05:42.42 | p3nguin | I'd love to see someone overcome that silly phone. |
| 05:42.54 | p3nguin | It would help a lot of other people. |
| 05:43.31 | K-Man`` | i will cetainly try...and ill post any solutions i find |
| 05:43.45 | K-Man`` | IF i find any that is |
| 05:48.10 | *** join/#asterisk jaxon007_ (~jayesh@123.252.144.92) |
| 06:03.52 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:05.55 | nitram | 7970 works fine with chan-sccp-b |
| 06:10.08 | K-Man`` | hi nitram... does this mean that you have to use a different image on the phone? or just enable a feature on asterisk? |
| 06:12.46 | *** join/#asterisk FainaUkraina (~Gene@059148208218.ctinets.com) |
| 06:12.58 | *** join/#asterisk asphalt (7514166e@gateway/web/freenode/ip.117.20.22.110) |
| 06:13.14 | asphalt | Hi Asterisk, is there any way I can get to know the order of events for Manager API? |
| 06:17.38 | WIMPy | puzzled: Do you have two versuons installed? One from the kernel and one in extra? |
| 06:18.07 | nitram | K-Man``: the image the phone usually comes with. SCCP/Skinny. |
| 06:19.16 | kaldemar | asphalt: not really. if the order gets changed in transport you'll just have to live with it. |
| 06:21.13 | *** join/#asterisk Brookss (~Gotenks_8@static-66-38-159-33.gtcust.grouptelecom.net) |
| 06:22.51 | nitram | K-Man``: running 7970, 7975, 7920, 7925, 7960, 7940 and 7937 with this |
| 06:23.02 | nitram | and the nokia icc |
| 06:23.48 | asphalt | kaldemar: How can the order be changed in transport? Isn't it TCP? |
| 06:24.16 | asphalt | kaldemar: Doesn't TCP guarantee order? |
| 06:24.39 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 06:26.25 | WIMPy | Has abyone had an issue with priority lables not being found? The thing that looks strange to me is that it seems to only affekt lables named "ok". |
| 06:26.53 | Brookss | tcp isn't an ideal voice protocol though... for data it is |
| 06:28.01 | asphalt | Brookss: Manager API protocol isn't voice |
| 06:28.30 | Brookss | oh got into the convo late |
| 06:30.42 | WIMPy | Ok. /me takes back the strange ok part. |
| 06:34.24 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 06:41.03 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:41.51 | K-Man`` | nitram.. sccp in a double nat type of environment? |
| 06:44.02 | kaldemar | asphalt: sure TCP does, but if you do something else with them too. |
| 06:44.31 | asphalt | kaldemar: What do you mean by 'something else with them'? |
| 06:46.11 | kaldemar | asphalt: anything that is not the TCP flow. |
| 06:47.11 | asphalt | kaldemar: Still not getting it? Isn't the Manager API only TCP? |
| 06:47.26 | asphalt | kaldemar: Still not getting it. Isn't the Manager API only TCP? |
| 06:48.57 | kaldemar | asphalt: yes. |
| 06:49.36 | kaldemar | asphalt: but if you have a bigger picture than the single TCP connection. sounds like you don't. |
| 06:55.11 | asphalt | kaldemar: You mean multiple ManagerConnections? |
| 06:55.31 | asphalt | kaldemar: Yep, then they won't be in order. |
| 06:56.04 | asphalt | kaldemar: Let's say I have only one ManagerConnection, does that guarantees the order of events? |
| 06:56.32 | asphalt | kaldemar: I mean does Asterisk send the events in order? |
| 06:57.06 | asphalt | kaldemar: For instance, will I receive the Leave (from queue) or Hangup (channel) event first? |
| 06:57.20 | asphalt | kaldemar: (assuming that the call was in queue.) |
| 06:59.32 | kaldemar | that's something you must test. |
| 07:04.06 | asphalt | kaldemar: Better would be to dive into the source. But it's a long task and testing wasn't giving good results (as the order varied in dev machine but in production machine it is different.) |
| 07:15.36 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:16.01 | *** join/#asterisk pietro (~pietro@88-149-226-83.dynamic.ngi.it) |
| 07:21.37 | autofsckk | how can i reload extensions.conf from cli? i forgot |
| 07:25.32 | *** join/#asterisk schmidts (~schmidts@213.235.212.196) |
| 07:25.34 | schmidts | good morning |
| 07:25.54 | kaldemar | autofsckk: "dialplan reload" |
| 07:26.51 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 07:27.25 | autofsckk | kaldemar: thanks |
| 07:27.58 | autofsckk | is it very dificult to have a conference room? i mean, the configuration |
| 07:28.33 | schmidts | autofsckk basically no ;) |
| 07:29.43 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 07:31.12 | autofsckk | :D |
| 07:31.38 | kaldemar | depends. it can be only one line in dialplan. |
| 07:32.39 | autofsckk | im reading the default file |
| 07:33.05 | autofsckk | im testing with a vps |
| 07:33.37 | kaldemar | which file? |
| 07:34.06 | autofsckk | meetme.conf.default |
| 07:36.32 | kaldemar | if you use option d or D for app MeetMe, you don't even need to configure meetme.conf. see "core show application MeetMe". |
| 07:37.16 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 07:39.46 | autofsckk | kaldemar: if i edit the meetme.conf file how can i load it? |
| 07:42.18 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 07:42.23 | autofsckk | oh i see, i have to put it on extensions.conf too right? |
| 07:42.35 | *** part/#asterisk Brookss (~Gotenks_8@static-66-38-159-33.gtcust.grouptelecom.net) |
| 07:43.04 | ChannelZ | Yeah |
| 07:43.06 | ChannelZ | it's not magic :) |
| 07:44.57 | ChannelZ | You said you were doing this on a VPS, note that you need DAHDI for MeetMe to work which may be a problem for you. You might be better off looking at ConfBridge assuming you're running a new-ish version of * |
| 07:47.52 | autofsckk | ChannelZ: i used a linode install with asterisknow, i deleted the freepbx part, but can you tell me please if theres a way to see if i have dahdi on the vps? |
| 07:49.20 | autofsckk | i mean, theres was an automated install, because i tried to install asterisk but i couldnt do it because of the need of kernel headers |
| 07:49.57 | kaldemar | autofsckk: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-7.html#meetmeConferencing |
| 08:08.49 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:09.41 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
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| 08:24.33 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:25.58 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 08:26.47 | *** join/#asterisk b0ot (~Jinxed---@212.183.140.3) |
| 08:27.22 | b0ot | Anyone ever have it where when you call a phone it just comes up congested? |
| 08:28.25 | kaldemar | b0ot: the call? the phone becomes congested? what phone? what call? |
| 08:28.38 | schmidts | is there any kind of "feature list" for asterisk available? |
| 08:28.50 | schmidts | basically i know about it but i need something "official" |
| 08:30.07 | b0ot | kaldemar, so when I call a phone and watch the cli I see it was congested |
| 08:30.13 | b0ot | and the phone never rings |
| 08:30.17 | b0ot | but will show a missed call |
| 08:30.35 | b0ot | however that phone shows ok in sip show peers |
| 08:30.39 | b0ot | and is able to call out |
| 08:31.22 | kaldemar | schmidts: http://www.asterisk.org/features |
| 08:31.36 | schmidts | kaldemar thanks i just found it ;) |
| 08:32.50 | kaldemar | b0ot: pastebin the output of "sip show peers" and the dialplan line that dials the phone. |
| 08:34.54 | b0ot | kaldemar, sip show peers |
| 08:34.55 | b0ot | http://paste2.org/p/1911188 |
| 08:35.20 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 08:36.18 | b0ot | kaldemar, extensions part: http://paste2.org/p/1911189 |
| 08:36.40 | b0ot | 8010/8010 192.168.0.30 D 5060 OK (55 ms) |
| 08:36.51 | b0ot | exten => 8010,1,Dial(SIP/8010,20) |
| 08:42.07 | kaldemar | pastebin is enough, you don't need to paste here too. give commands "core set verbose 10", "sip set debug on" and make a call. then pastebin everything you get in CLI. |
| 08:42.15 | b0ot | ok |
| 08:42.18 | ChannelZ | can you show the console output when you dial it? Seems like something else has to be at play. Or it's fine and the device is rejecting the call specifically |
| 08:43.16 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:44.02 | b0ot | call before sip debug for call |
| 08:44.03 | b0ot | http://paste2.org/p/1911195 |
| 08:44.41 | kaldemar | "-- Got SIP response 603 "Decline" back from 192.168.0.30" |
| 08:47.10 | kaldemar | that usually happens when the callee hangs up the call (i.e. rejects) before answering. |
| 08:51.32 | b0ot | kaldemar, the phone shouldn't be rejecting it |
| 08:51.39 | b0ot | I can see it... it just is sitting there |
| 08:51.51 | b0ot | I have reset the phone multiple time |
| 08:51.54 | b0ot | times* |
| 08:52.34 | ChannelZ | Does it have a DND function? |
| 08:53.12 | WIMPy | Is it normal that DND results in congetion? |
| 08:53.20 | kaldemar | DND usually replies with a message different from 603. |
| 08:53.23 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 08:53.39 | ChannelZ | Retard phone? Who knows. |
| 08:54.03 | kaldemar | DND usually results to 480 or 486. |
| 08:54.16 | WIMPy | I do indeed get congestion when I activate DND. |
| 08:54.48 | WIMPy | But I filed that under "perfectely normal SIP incompatibilities". |
| 08:55.04 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 08:55.54 | b0ot | kaldemar, http://paste2.org/p/1911201 the call |
| 08:56.05 | b0ot | I have used this phone before and was able to call it no problem |
| 08:56.23 | b0ot | sip debug output of call |
| 08:56.25 | kaldemar | the congestion word is just a general dumb indication that something went wrong. |
| 08:56.54 | WIMPy | "wrong"? |
| 08:57.20 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 08:58.10 | ChannelZ | Found peer '9005' for '8001' from 192.168.0.150:5060 ?? |
| 08:59.17 | kaldemar | WIMPy: exactly. the "Everyone is busy/congested at this time" is just a general indication that doesn't say much about what really happened. |
| 09:00.24 | WIMPy | "general failure" |
| 09:00.44 | WIMPy | As in "Who is General Failure? And why is he reading my disk?"? |
| 09:01.20 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:01.52 | kaldemar | b0ot: maybe you've enabled DND since then. check it. |
| 09:01.53 | b0ot | sip.conf http://paste2.org/p/1911203 extenstions.conf http://paste2.org/p/1911204 |
| 09:01.59 | b0ot | on the phone? |
| 09:02.06 | kaldemar | b0ot: yes, on the phone. |
| 09:02.19 | b0ot | hahahaha |
| 09:02.20 | b0ot | damn |
| 09:03.02 | b0ot | problem solved |
| 09:03.03 | ChannelZ | Thanks for listening |
| 09:06.25 | *** join/#asterisk g011um (~fred.taff@LLagny-156-35-17-119.w80-14.abo.wanadoo.fr) |
| 09:06.39 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 09:06.50 | g011um | hi all |
| 09:07.08 | g011um | question about dynmeetme |
| 09:07.29 | g011um | if someone try to join the meetme when a people is recording his greating message, he receive a busy tone |
| 09:08.29 | ChannelZ | I'd guess because the conference really doesn't exist yet.. |
| 09:08.39 | g011um | hanhan |
| 09:09.03 | g011um | in fact it's a dynmeetme. The conference already exist |
| 09:10.29 | g011um | when the user join the meetme, he record his name and valid by the pound key |
| 09:10.53 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:11.02 | g011um | if at the same time another try to call the meetme number he obtain a busy tone |
| 09:11.12 | ChannelZ | but is this person the first one in the conference? |
| 09:11.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:11.38 | g011um | no the meetme room is already opened |
| 09:11.40 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:14.01 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 09:15.11 | *** join/#asterisk g011um (~fred.taff@LLagny-156-35-17-119.w80-14.abo.wanadoo.fr) |
| 09:15.18 | g011um | oups sorry |
| 09:41.55 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.88.157.dsl.dyn.forthnet.gr) |
| 09:52.07 | *** join/#asterisk dandate2 (~dan@180.190.222.190) |
| 09:52.19 | dandate2 | does anyone know where i can find an example .conf file for the cisco 7912 ip phone? |
| 09:58.21 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:00.24 | dandate2 | or the firmware |
| 10:04.17 | *** join/#asterisk schmidts (~schmidts@213.235.212.196) |
| 10:24.28 | *** join/#asterisk RickCogley_ (~anonymous@114.179.8.185) |
| 10:28.49 | *** join/#asterisk g011um (~fred.taff@LLagny-156-35-17-119.w80-14.abo.wanadoo.fr) |
| 10:29.46 | g011um | good morning |
| 10:30.36 | g011um | do you know wich config file defined the number of internal voice channel ? |
| 10:32.48 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 10:33.01 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 10:33.22 | g011um | i have an issue with dynemmetme. could'you help me ? |
| 10:35.03 | schmidts | g011um what you mean with internal voice channels? there is not such a limit |
| 10:35.24 | g011um | ok here's my issue : |
| 10:35.56 | g011um | i use a number to create dynmeetme and another to join the meetme |
| 10:36.13 | g011um | first, i create a meetme |
| 10:36.20 | kaldemar | how? |
| 10:36.42 | kaldemar | pastebin your relevant dialplan. |
| 10:37.06 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:37.07 | g011um | by calling the number to create the dynmeetme then enter a number of meetme. |
| 10:38.15 | g011um | if a people is recording his meetme greating while joining the dynmeetme, a second one calling in the same time obtain a busy tone |
| 10:38.24 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 10:41.13 | schmidts | g011um this shouldnt happen but as kaldemar said, please pastebin your relevant dialplan |
| 10:44.57 | saxa | [Feb 21 11:41:15] WARNING[8646]: channel.c:5104 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin) |
| 10:45.10 | saxa | what exactly this means ? |
| 10:45.24 | *** join/#asterisk Donper (~donperx@sto93-2-82-228-142-248.fbx.proxad.net) |
| 10:45.37 | saxa | http://pastebin.com/ySbj0spi |
| 10:46.24 | kaldemar | saxa: it means that you don't have a g.729 codec. |
| 10:46.38 | saxa | but why it tries to use it ? |
| 10:46.54 | saxa | i never specified that codec |
| 10:47.04 | saxa | ok let me investigate |
| 10:49.11 | kaldemar | saxa: ludmila tries to use g.729- |
| 10:49.16 | saxa | kaldemar: and what about the line 43 and 44 in the apste ? |
| 10:49.25 | saxa | any idea ? |
| 10:49.41 | g011um | http://pastebin.com/5HScPYP9 |
| 10:49.44 | g011um | thanks |
| 10:50.48 | kaldemar | saxa: ludmila tries to use g.729 and you don't have a codec to transcode between that and what DAHDI wants. |
| 10:56.24 | saxa | kaldemar: ok, I had my-codecs macro in use on ludmila sip phone. So changed that to ulaw-codec |
| 10:56.36 | saxa | let me try again to see what happens. |
| 10:59.33 | saxa | heh |
| 11:00.11 | saxa | now using the ulaw on ludmila works, i can even call out from it over dahdi and I can hear the other side |
| 11:00.25 | saxa | this is not even close to do it with sasa |
| 11:00.39 | saxa | both have exactly the same setting in sip.conf |
| 11:01.15 | kaldemar | but en entirely different error in the CLI. |
| 11:01.33 | saxa | so both use the ulaw codec and everything, one is Yealink sip-t32g (ludmila) which works, and the other one is Yealink sip-t38g (sasa) which does not work. |
| 11:01.37 | kaldemar | sasa dialed SIP/1002 => "Purely numeric hostname (1002), and not a peer--rejecting!" |
| 11:01.54 | saxa | kaldemar: let me paste the last lines |
| 11:02.10 | saxa | this was before i changed the sip.conf for ludmila |
| 11:02.25 | saxa | but yes, I still cant call ludmila from sasa |
| 11:03.07 | saxa | http://pastebin.com/6pMcs1HX here is it. |
| 11:03.15 | saxa | thats after a sip relaod |
| 11:04.52 | kaldemar | you're not even trying to dial ludmila. |
| 11:04.56 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:05.05 | saxa | ok, i fixed that |
| 11:05.31 | saxa | i changed the extensions.conf to dial SIP/ludmila and not SIP/1002 |
| 11:05.54 | saxa | now it rings on the other sire, so sasa can call ludmila now. |
| 11:06.59 | saxa | I had in my extensions.conf a wrong setting in a LUDMILA variable. |
| 11:07.02 | dandate2 | anyway to quickly get a CCO login for a phone firmware, this whole having to wait for a reseller to call me so i can buy 1 freaking license is trauma |
| 11:10.16 | saxa | ok the thing now is, ludmilas phone works perfectly, sasas no. |
| 11:10.34 | saxa | so ludmila can call ou via dahdi and hear, sasa not. |
| 11:11.18 | saxa | on sasas phone it seems that asterisk never connects dahdi channel to sip |
| 11:12.26 | saxa | because on the sasa phone when i call out i continue to see the dialling out arrows moving on the right (a kind of progrss bar) also when i have already answered a call on my cell phone. |
| 11:12.45 | saxa | on ludmilas phone doesnt happen this. |
| 11:13.23 | saxa | when i answer, the progress bar stops and it shows the full display. |
| 11:13.28 | g011um | please, could'you watch on my pastbin ? http://pastebin.com/5HScPYP9 |
| 11:13.43 | saxa | i looked at it g011um |
| 11:13.50 | saxa | but i don't know whats on |
| 11:14.59 | kaldemar | saxa: look at asterisk when you dial, it tells more than a progress bar. |
| 11:15.31 | saxa | yes |
| 11:15.39 | saxa | it seems nothing wrong there |
| 11:16.23 | kaldemar | g011um: nothing wrong there. what do you get in CLI when it happens? |
| 11:17.06 | saxa | http://pastebin.com/prVyGSLm <-- this is the console out with also a rtp set debug on during the call and a sip set debug on. |
| 11:17.37 | g011um | it's a real difficulty to trace this issue. Because of the use of our asterisk |
| 11:18.26 | g011um | we have two elastix : one for voicemail and another one for fax and meetme |
| 11:18.42 | saxa | ok this rtp and sip debus are for the call from ludmila. Probably useless. |
| 11:19.03 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 11:19.05 | g011um | these two asterisk are linked to our avaya s8700 (2 h323 trunks) |
| 11:19.47 | g011um | so when i lauch asterisk -r there's alot of traffic |
| 11:20.15 | g011um | but do you no if it could be an issue with dahdi pseudo channels ? |
| 11:20.29 | saxa | http://pastebin.com/Ne4ZKSkR <-- here is the call from sasa to my cell. |
| 11:22.53 | kaldemar | saxa: "-- DAHDI/1-1 is busy" |
| 11:23.14 | saxa | http://pastebin.com/zXz7358b |
| 11:23.22 | kaldemar | saxa: either something already uses your channel 1 or the destination is busy. |
| 11:23.27 | saxa | kaldemar: that happens when i answer the phone |
| 11:23.35 | g011um | do you know how to monitor dahdi usage ? |
| 11:24.06 | saxa | when I answer the phone the console shows dahdi is busy |
| 11:24.30 | kaldemar | saxa: what is DAHDI/1? |
| 11:26.05 | saxa | oh I looked at it right now closer, basically when i dial out from the sip phone it rings and the last message on the console is |
| 11:26.08 | saxa | <PROTECTED> |
| 11:26.15 | saxa | it stays there |
| 11:26.24 | *** join/#asterisk Srini (Srinivasa@116.202.141.190) |
| 11:26.29 | saxa | after i answer the cell phone it still sits there |
| 11:27.01 | saxa | then i disconnect the cell phone and I get the following messages: |
| 11:27.05 | saxa | <PROTECTED> |
| 11:27.05 | saxa | <PROTECTED> |
| 11:27.05 | saxa | <PROTECTED> |
| 11:27.05 | saxa | <PROTECTED> |
| 11:27.05 | saxa | <PROTECTED> |
| 11:27.08 | saxa | <PROTECTED> |
| 11:27.11 | saxa | quadser*CLI> |
| 11:27.40 | saxa | kaldemar: DAHDI/1 is the TDM410P |
| 11:27.45 | saxa | first line |
| 11:27.52 | *** join/#asterisk barfoo365 (~barfoo365@cpc1-stav6-0-0-cust1454.17-3.cable.virginmedia.com) |
| 11:27.59 | saxa | basically the only one connected right now |
| 11:28.31 | *** join/#asterisk gavimobile (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 11:28.53 | kaldemar | so it's an analog line. and you can successfully dial out the same channel and number with the other phone? |
| 11:29.49 | saxa | http://pastebin.com/tUAH870B |
| 11:30.04 | saxa | yes kaldemar , here you see a call from DAHDI/3 my analog phone |
| 11:30.25 | saxa | kaldemar: yes, if I call from ludmila to my cell phone I hear everything |
| 11:31.02 | saxa | if i call from sasa , i do hear the ringing on my cell phone, and when i answer there is no audio at all. |
| 11:31.13 | gavimobile | I need some direction here please. 2 users each have 2 trunks from the same provider on the same server connecting to the same server. my trunks do not need username or secret to connect to the provider. how does my provider know which user to bill? |
| 11:32.04 | saxa | kaldemar: on ludmila for example, when i dial out to my cell phone, i see that progress bar, and it disapears immediately after i answer the cell phone. |
| 11:32.17 | kaldemar | saxa: i can't think of anything but a codec issue. |
| 11:32.18 | saxa | kaldemar: thats not the case for sasa phone |
| 11:32.45 | saxa | strangely, both phones are using the same sip config ulaw |
| 11:33.05 | kaldemar | saxa: care to show it? |
| 11:33.12 | saxa | of course no problem |
| 11:33.30 | saxa | let me just grep -v \; the comments |
| 11:34.33 | kaldemar | then show a whole call from sasa with sip debug enabled. |
| 11:35.44 | kaldemar | gavimobile: by caller id for example. |
| 11:36.20 | saxa | http://pastebin.com/4PN7BeXn |
| 11:37.14 | gavimobile | can 2 different people use the same trunk information on the same pbx server? |
| 11:37.41 | gavimobile | kaldemar: I don't think the provider bills according to the caller id, because if I change my caller id to something else I would still get billed |
| 11:37.57 | kaldemar | saxa: your paste hides [basic-options] and [ulaw-phone]. also, you should not have configurations directly under [authentication]. |
| 11:38.15 | saxa | http://pastebin.com/uWRVa5Mj |
| 11:38.24 | saxa | here is the sip debug on |
| 11:38.54 | saxa | kaldemar: maybe i deleted it |
| 11:38.59 | saxa | let me recheck |
| 11:39.14 | kaldemar | saxa: the grep -v \; left important pieces out. |
| 11:39.14 | saxa | because there was plenty of white lines so i make it smaller |
| 11:41.23 | saxa | i'm trying to paste the complete sip.conf, but it freezed my firefox :D |
| 11:41.27 | gavimobile | how do providers identify which channel was used for outgoing calls? |
| 11:42.39 | kaldemar | saxa: sed < sip.conf -e '/^\s*;/d' -e '/^$/d' |
| 11:43.09 | kaldemar | gavimobile: maybe you should ask your provider. |
| 11:44.06 | gavimobile | kaldemar: thanks |
| 11:46.16 | *** join/#asterisk RickCogley (~anonymous@EM114-49-5-124.pool.e-mobile.ne.jp) |
| 11:48.09 | barfoo365 | Can anyone see why my SIP phone is not answering the call when I pick it up? http://pastebin.com/MtHJ8nn1 |
| 11:50.01 | saxa | still freezed :) |
| 11:50.15 | WIMPy | barfoo365: No, But I can see that it did answer. |
| 11:50.58 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 11:51.16 | barfoo365 | It did? Strange. Basically I am calling in to a POTS line, the call is correctly routed to my SIP phone but when I pick it up i just hear nothing. And my cellphone is still ringing |
| 11:51.41 | elliot98 | does the RTP SSRC show up anywhere in SIP packets? |
| 11:52.57 | barfoo365 | sorry elliot98, im a bit lost :) |
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| 11:58.18 | schmidts | elliot98 RTCP normally uses another port than sip |
| 11:58.18 | elliot98 | barfoo365: in other words how does one associated a SIP connection with its RTP packets |
| 11:59.06 | saxa | kaldemar: sorry i have to go out, will be back later, the thing is still frozen, huh |
| 12:01.55 | barfoo365 | Now ive got an even stranger issue happening, the phone only seems to work once. After one attempt at a call I have to unplug it and then back in for it to work again |
| 12:01.56 | barfoo365 | strange |
| 12:05.15 | elliot98 | schmidts: aside from the port and ip in an INVITE packet, what else is there to associate the SIP call with the RTP packets? |
| 12:06.05 | *** join/#asterisk BenC[UK] (~bcummins@host90-152-2-82.ipv4.regusnet.com) |
| 12:06.11 | BenC[UK] | Hi Guys |
| 12:06.16 | BenC[UK] | I've just set up a new server |
| 12:06.26 | BenC[UK] | and takign complete config from one server thats working fine |
| 12:06.31 | BenC[UK] | but on new server having iss.. WARNING[27791] channel.c: No path to translate from SIP/voipms-00000003 to Local/DIALCUSTOMER@DIALPLAN-1-1d12;2 |
| 12:08.03 | schmidts | elliot98 the sdp information, but the SSRC is computed of the source ip + port IMHO and does not belong to sip |
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| 12:32.56 | *** join/#asterisk mahaD (mahaD@14.99.21.190) |
| 12:33.43 | mahaD | hello all, does cdr-stats support adaptive mysql odbc cdr also ? |
| 12:34.54 | elliot98 | getting a strange situation that the UA does not start sending RTP packets even after getting a 200 response |
| 12:36.42 | schmidts | elliot98 maybe a network problem? |
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| 12:41.29 | mahaD | in cdr_adaptive_odbc.conf , if we use filter accountcode > myac , then that will log only those calls, right ? or will it filter out those calls in the said cdr table? |
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| 12:53.39 | BenC[UK] | was codec issue, thanks guys |
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| 13:13.41 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 13:17.55 | Srini | hi room |
| 13:18.41 | Srini | I am still lost in the DAHDI/g1/${EXTEN}, my calls or not going out... I still get the extension not found error :( |
| 13:20.23 | [TK]D-Fender | Srini, pastebin <- |
| 13:20.25 | [TK]D-Fender | ~pb |
| 13:20.25 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:21.30 | Srini | http://pastie.org/3426944 |
| 13:22.24 | *** join/#asterisk schmidts (~schmidts@213.235.212.196) |
| 13:22.43 | [TK]D-Fender | Srini, And the failed call with full debug... |
| 13:23.20 | [TK]D-Fender | Srini, Line 5 in there is a duplication and should eb removed |
| 13:23.24 | [TK]D-Fender | be* |
| 13:23.50 | [TK]D-Fender | Srini, outgoing ALSO has a complete duplication of the sme pattern. remove these duplicates. |
| 13:24.14 | kaldemar | Srini: "#/etc/asterisk/extension.conf" extension.conf or extensions.conf? |
| 13:25.01 | [TK]D-Fender | Yup, wrong file altogether |
| 13:32.08 | barfoo365 | [TK]D-Fender : i'm back :) Remember me from yesterday? |
| 13:33.28 | Srini | [TK]D-Fender, it is extensions.conf |
| 13:34.00 | [TK]D-Fender | Srini, Where is the call & the corrected config? |
| 13:37.49 | elliot98 | schmidts: It works fine if progressinband is set up, basically if the system sends 183, the RTP packet transaction starts, but sending a 180, nothing is sent from the UA after the 200 response |
| 13:38.14 | elliot98 | schmidts: so it wouldn't seem to be a network problem |
| 13:39.19 | Srini | [TK]D-Fender, I have updated the pastie: http://pastie.org/3426944 |
| 13:39.57 | [TK]D-Fender | Srini, what part of "full debug" was not clear? Enable SIP DEBUG and do it again |
| 13:40.10 | Srini | ok |
| 13:40.30 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 13:43.47 | Srini | [TK]D-Fender, http://pastie.org/3427044 |
| 13:44.10 | [TK]D-Fender | Srini, Looking for 9886065975 in default (domain 115.115.80.125) <------- |
| 13:44.22 | [TK]D-Fender | Srini, default != outgoing |
| 13:44.27 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:45.04 | Srini | [TK]D-Fender, so the context problem? |
| 13:45.08 | [TK]D-Fender | clearly |
| 13:45.24 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:45.40 | Srini | so it is again in chan_dahdi.conf that I have to check for the outgoing context? |
| 13:45.52 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:46.10 | Srini | where do I set the context for the outgoings? |
| 13:47.25 | [TK]D-Fender | sirion your PHONE <_ |
| 13:47.28 | [TK]D-Fender | SIP.CONF |
| 13:47.35 | [TK]D-Fender | You aren't looking where your PHONE is pointed |
| 13:47.53 | [TK]D-Fender | Srini, that call has nothing to do with DAHDI |
| 13:52.08 | *** join/#asterisk mintos (~mvaliyav@114.143.161.91) |
| 13:52.18 | Srini | [TK]D-Fender, That helped! |
| 13:53.35 | autofsckk | i get -- Got SIP response 603 "Declined" back from how can i fix that? |
| 13:53.50 | autofsckk | == Everyone is busy/congested at this time (1:1/0/0) |
| 13:53.57 | kaldemar | autofsckk: does your phone have DND on? |
| 13:55.25 | [TK]D-Fender | Do we even know it's a "phone" that he's talking to? |
| 13:55.34 | autofsckk | im reading about DND atl voip-info.org, but i dont think so, this is a new asterisk that im testing, trying to make a call to my cell phone |
| 13:55.56 | [TK]D-Fender | autofsckk, we have not idea what you are calling to. Show us the call with SIP DEBUG |
| 13:56.03 | [TK]D-Fender | ~pb |
| 13:56.03 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:56.05 | [TK]D-Fender | ^^^ |
| 13:56.53 | autofsckk | [TK]D-Fender: can you remind me please how can i make a sip debug to a file so i can paste the content? |
| 13:57.09 | schmidts | autofsckk sip set debug peer xyz |
| 13:57.15 | [TK]D-Fender | * CLI -> "sip set debug on" |
| 13:57.19 | [TK]D-Fender | Copy. Paste. |
| 13:57.26 | [TK]D-Fender | NO. |
| 13:57.39 | [TK]D-Fender | Don't just limit to 1 peer, We need to see BOTH sides of things |
| 13:57.45 | schmidts | ok sorry ;) |
| 14:01.03 | barfoo365 | Dialing in from an outside line the phone rings once then stops and shows missed call, then rings again and so on http://pastebin.com/ZArnhJDQ Anyone any ideas whats happening? |
| 14:04.15 | barfoo365 | Im assuming some sort of call timeout/ |
| 14:04.54 | [TK]D-Fender | barthere is a spot in general settings to define it. go fix this. Also... |
| 14:04.56 | [TK]D-Fender | ~freepbx |
| 14:04.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:04.58 | [TK]D-Fender | ^^^ |
| 14:05.57 | *** join/#asterisk serafie1 (~erin@nat/digium/x-rmmcsylosmypbpjb) |
| 14:06.10 | Srini | [TK]D-Fender, Thanks a lot! Things worked like charm after that! |
| 14:06.26 | Srini | Thanks room for all the help! |
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| 14:42.59 | barfoo365 | I have found this bug and patch which appears to relate to my issue https://issues.asterisk.org/view.php?id=18667 |
| 14:43.22 | barfoo365 | Can I just apply to my asterisk install or do i need to apply it to source then recompile? |
| 14:44.09 | Gugge | barfoo365: you could just upgrade to an asterisk version newer than 2011-03-01 |
| 14:46.30 | barfoo365 | I am using version 1.8.6.0 |
| 14:46.35 | barfoo365 | which appears to be August 2011 |
| 14:46.47 | barfoo365 | so presumably that patch should already be applied then? |
| 14:50.33 | kaldemar | barfoo365: seems to be. |
| 14:51.39 | *** join/#asterisk emate (~marcin@178-37-96-34.adsl.inetia.pl) |
| 14:51.42 | emate | hi there |
| 14:54.28 | emate | i have one question, how can i group all actions taken for incoming call? (ex. Call goes into queue -> asterisk calls extension aaa (NO ANSWER) -> asterisk calls extension bbb (NO ANSWER) -> asterisk call extension ccc (ANSWERED)) |
| 14:55.25 | emate | i have uniqueid, but it sometime differs between switched calls (but logicaly it's still one incoming connection). |
| 15:03.52 | saxa | kaldemar: I finally go it posted :D |
| 15:03.55 | saxa | http://pastebin.com/FyzFAz2p |
| 15:04.21 | saxa | sorry i was out, so here is the complete sip.conf where you can see the missing options |
| 15:06.35 | [TK]D-Fender | saxa, Permanently remove all that commented out garbage |
| 15:06.58 | saxa | :) |
| 15:07.17 | saxa | I know, let me use kaldemar's sed |
| 15:08.36 | saxa | http://pastebin.com/QN5e6t0h <-- here we go |
| 15:09.32 | [TK]D-Fender | saxa, So since it has been many hours... why are you showing this? |
| 15:10.14 | saxa | [TK]D-Fender: kaldemar said, that wanted to see my sip.conf, since he think there is a codec issue |
| 15:10.32 | saxa | I posted before also the sip debug |
| 15:10.35 | [TK]D-Fender | saxa, Show us the call you suspect this is for. |
| 15:10.46 | saxa | let me pick that up |
| 15:10.51 | [TK]D-Fender | saxa, "Before" was many hourse ago on a different client |
| 15:11.19 | saxa | http://pastebin.com/uWRVa5Mj |
| 15:11.41 | saxa | [TK]D-Fender: no its still on the same phone, i was out |
| 15:11.47 | saxa | so i get back right now |
| 15:12.03 | [TK]D-Fender | saxa, - DAHDI/1-1 is busy <------- BUSY |
| 15:12.04 | saxa | the phone in the question is the sasa |
| 15:12.11 | [TK]D-Fender | where do you get the idea this is a codec issue? |
| 15:12.28 | saxa | [TK]D-Fender: kaldemar said that |
| 15:12.33 | kaldemar | looks like it's something else, there is a common codec. |
| 15:12.47 | saxa | yes thats what I think too |
| 15:12.57 | saxa | the other phone works with this same codec |
| 15:13.12 | [TK]D-Fender | The call is ulaw (broing) and the call was never answered. there is no chance for a conflict |
| 15:13.23 | saxa | I can call DAHDI/3 to DAHDI/1 , I get connected |
| 15:13.25 | [TK]D-Fender | saxa, show us this other call |
| 15:13.40 | [TK]D-Fender | DAHDI/1-1 is busy <------- BUSY |
| 15:13.42 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:13.45 | kaldemar | [TK]D-Fender: in short, "ludmila" can dial out of DAHDI/1 successfully, but "sasa" has no voice. |
| 15:13.54 | saxa | they all get DAHDI/1-1 is busy |
| 15:14.25 | saxa | this DAHDI/1-1 is busy appears only when I disconnect the phone |
| 15:14.32 | [TK]D-Fender | "no voice"? |
| 15:14.37 | saxa | nothing |
| 15:14.41 | saxa | no audio |
| 15:14.44 | [TK]D-Fender | what do you mean? |
| 15:14.54 | [TK]D-Fender | IIt looks like it dies immediately |
| 15:15.15 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:15.15 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:15.16 | [TK]D-Fender | Show us this other call |
| 15:15.38 | barfoo365 | Can anyone see anything out of the ordinary in this http://pastebin.com/MY4x7RMe It seems to be detecting that I have picked the phone up but the other end just keeps ringing |
| 15:15.48 | saxa | [TK]D-Fender: I see on sasa that when I call out, it shows a progress bar and it never changes to an answered state, so this progress bar continues, which is not true for ludmila. |
| 15:15.53 | kaldemar | also, DAHDI/1 is analog. |
| 15:16.03 | [TK]D-Fender | Show us the other call |
| 15:16.04 | saxa | ok let me do a sip debug on on ludmila |
| 15:17.15 | [TK]D-Fender | Same call from other device. |
| 15:20.37 | saxa | http://pastebin.com/nMzjpPhY |
| 15:20.47 | saxa | ^^^ this is a call from ludmila |
| 15:21.13 | saxa | I had to do it 3 times since my mobile was not on the providers network |
| 15:21.17 | saxa | bad signal |
| 15:21.40 | saxa | so the first 2 attempts are answering machine |
| 15:21.59 | saxa | but no matter, the thing is that I can hear everything on ludmila |
| 15:22.14 | saxa | everything coming from DAHDI or coming from SIP |
| 15:22.51 | saxa | and you can also see -- DAHDI/1-1 is busy |
| 15:23.45 | [TK]D-Fender | asxI don't see this second call getting answered or ringing or anything... |
| 15:23.55 | saxa | ?? |
| 15:24.00 | saxa | it answered it |
| 15:24.05 | saxa | and heard my voice |
| 15:25.25 | [TK]D-Fender | I don't see an answer.... |
| 15:26.00 | saxa | ok so what should I do ? |
| 15:26.08 | [TK]D-Fender | another call |
| 15:26.11 | saxa | ok |
| 15:26.14 | saxa | let me do it |
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| 15:32.35 | saxa | http://pastebin.com/E04mBdFH |
| 15:32.48 | saxa | Ok this one is another call |
| 15:33.32 | *** part/#asterisk Srini (Srinivasa@116.202.141.190) |
| 15:34.07 | saxa | is there a dahdi set debug on option ? |
| 15:36.59 | kaldemar | saxa: not for analog. "core set debug 10" might show something useful if you have "debug" on the console line in logger.conf. |
| 15:37.16 | saxa | i issued that |
| 15:37.44 | saxa | oh ok, maybe i dont have debug on the console in logger.conf |
| 15:37.54 | saxa | let me check |
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| 15:40.12 | lirakis | we've got some old asterisk 1.2 gateway boxes that interface via SIP for termination |
| 15:40.24 | lirakis | one of our termination providers is doing bogus fax detection |
| 15:40.30 | lirakis | and sending a t.38 reinvite |
| 15:40.34 | lirakis | after the call has setup |
| 15:40.38 | lirakis | obv. this doesnt work |
| 15:40.45 | lirakis | b/c 1.2 has no t.38 support |
| 15:41.32 | lirakis | any how, asterisk tries to process_sdp() and hangs. eventually the process that recieved the reinvite crashes. |
| 15:41.51 | lirakis | any way to "disable" t.38 reinvite processing on asterisk 1.2? |
| 15:42.03 | lirakis | so that they are just ... rejected |
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| 15:43.11 | tzafrir | If I build dahdi from a version that is not the latest, and chance of it pulling an incorrect version of the Digium EC firmware ? |
| 15:43.47 | tzafrir | (helping someone, and the question is due to something that appears to have been lost in translation) |
| 15:45.09 | barfoo365 | Im still no further to resolving my issue, for some reason when an incoming call is detected, asterisk appears to be flaky on the incoming call detection - http://pastebin.com/EJX3duWV |
| 15:45.19 | tzafrir | That is: will it pull the "right version" or the "latest version"? Is there any chance it will pull a version that is incompatible? |
| 15:45.53 | pigpen | tzafrir, you know, I have only done upgrading, not downgrading. |
| 15:45.55 | saxa | [TK]D-Fender: have you saw the seccond pastebin ? |
| 15:46.19 | pigpen | that being said, I did go from 1.8 to 1.6 without issues on a box that had both analog and digital cards in it. (digium) |
| 15:46.21 | pigpen | with no issue. |
| 15:46.35 | [TK]D-Fender | saxa, I am still never seeing it answered |
| 15:47.35 | saxa | but as i said, i hear the voice and I can talk with the other side |
| 15:47.45 | saxa | [TK]D-Fender: thats probalby the cause of the problem |
| 15:48.26 | pigpen | I know this is a shot in the dark, with heavy wind?.but ? Anyone setup an audiocodes with t.38 support, terminating on asterisk utilizing res_fax_digium? |
| 15:51.43 | saxa | http://pastebin.com/0WZce1RQ <-- here is my dahdi-channels.conf and chan-dahdi.conf |
| 15:53.41 | *** part/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
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| 15:54.59 | saxa | [TK]D-Fender: do you want me to do one call from DAHDI/3 ? |
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| 15:56.02 | [TK]D-Fender | saxa, No.... we are comparing tests ont he same channel.. You want the environment to be controlled to find out what the difference is |
| 15:56.17 | [TK]D-Fender | saxa, And I'm really not seeing it right now.... |
| 15:56.30 | saxa | http://pastebin.com/CDykdF6B |
| 15:57.12 | saxa | ok here is my DAHDI/3 (analog phone on same TDM410 card. |
| 15:57.16 | saxa | ops |
| 15:57.25 | saxa | oh ok not needed |
| 15:57.46 | saxa | anyway, as you see here I also get this DAHDI/1-1 is busy |
| 15:58.41 | fireman_biff | Hi, what should I do if a PRI is showing as down even though there are no alarms and dahdi_tool shows it as OK? When debugging the span I see this line fairly often: "PRI Span: 1 TEI=0 Sending SABME" and occasionally see this line: "PRI Span: 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment)" |
| 15:58.51 | fireman_biff | asterisk 1.8.7.0 |
| 15:59.05 | [TK]D-Fender | saxa, I'm wondering about yoru indications zones.. try this : "callprogress=no" for your channels. |
| 15:59.08 | saxa | I call out, then it rings on my cell phone and until i disconnect on the cell phone it stays on the Called..... (line nr.5), only when I disconnect it apears the DAHDI/1-1 is busy. |
| 15:59.10 | [TK]D-Fender | Retest, see if that helps |
| 15:59.29 | saxa | on all 3 channels ? |
| 15:59.52 | saxa | in dahdi-channels.conf correct ? |
| 16:01.04 | saxa | oh its in chan_dahdi.conf |
| 16:01.41 | [TK]D-Fender | just port 1 |
| 16:01.47 | [TK]D-Fender | restrict your testing environment |
| 16:03.10 | saxa | ok I will add that there. |
| 16:03.29 | saxa | [TK]D-Fender: do you want then a new sip set debug on from ludmila, correct? |
| 16:03.51 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 16:03.55 | [TK]D-Fender | saxa, just test yourself with both first.... then do another with more debug if you feel like showing |
| 16:04.07 | [TK]D-Fender | saxa, I'm reaching the end of my inspiration on this.... |
| 16:04.21 | [TK]D-Fender | saxa, You will have to restart * for that change |
| 16:05.09 | *** join/#asterisk Kroms (~Kroms@72.37.252.50) |
| 16:06.46 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:06.53 | saxa | http://pastebin.com/KVknghFk |
| 16:07.03 | saxa | there is still no answer in my opinion |
| 16:07.27 | saxa | since I see now on ludmilas phone also a progress bar during the answered call |
| 16:07.44 | saxa | this means that the phone is still thinking it is waiting for the connection |
| 16:07.57 | saxa | but in fact i can hear myself talking on the phone |
| 16:08.50 | saxa | [TK]D-Fender: i did, I will try to restart over the chan-dahdi.conf I think |
| 16:09.32 | saxa | i set up yesterday a minimal dialplan and a sip.conf, basically no options in there, but i still have gotten no audio |
| 16:09.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 16:09.50 | saxa | so I think the problem is on the dahdi side |
| 16:11.02 | [TK]D-Fender | Possible... |
| 16:11.20 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:11.25 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 16:11.27 | [TK]D-Fender | eek |
| 16:11.45 | saxa | ok if you have any other suggestions what should i look are very welcome :D |
| 16:15.12 | *** join/#asterisk lal00 (~eduardo@201.151.40.230) |
| 16:16.02 | lal00 | I'm having some troubles with DTMFs, not all of the keypresses are being identified by asterisk. This happens randomly. Any idea what to check/debug? |
| 16:17.05 | [TK]D-Fender | saxa, So no improvement/change? |
| 16:17.45 | *** part/#asterisk lirakis (~lirakis@ool-45752d29.dyn.optonline.net) |
| 16:19.18 | *** join/#asterisk emate (~marcin@static-81-219-63-19.devs.futuro.pl) |
| 16:21.21 | *** join/#asterisk mentax (~mentax@rrcs-24-103-48-205.nyc.biz.rr.com) |
| 16:21.28 | mentax | hi all |
| 16:21.55 | emate | guys, can i set some global unique variable to incoming call, so i can identify this connection later in system? |
| 16:22.46 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-183.mobile.uci.edu) |
| 16:22.52 | tzanger | any E1 experts here? I'm trying to find the jitter specification for E1 clocks |
| 16:22.53 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:23.07 | _Corey_ | emate: "global" != unique to call ... you can set variables as needed that will be available for the call's duration, or global variables that will persist |
| 16:23.27 | tzanger | I now that 2.048MHz is a 433ns period, but how much jitter is acceptable as an input to an E1 framer |
| 16:25.16 | pigpen | lal00, how are you getting service? dahdi, sip, iax? |
| 16:25.27 | lal00 | pigpen: sip |
| 16:25.47 | pigpen | lal00, from a provider or a sip gateway? |
| 16:26.08 | lal00 | from a gateway |
| 16:26.24 | mentax | have a problem, when I call to my asterisk from cell phone, I hear the dialtone in my cell, but when I trying to call from landline - I don't hear dialtone in my land line phone. My extension ringing anyway, but the problem is people who call to my phone using land line doesn't here dial tone, If I was not answering, it send them to voicemail... |
| 16:26.27 | pigpen | what what is the gateway? Also, what is the client producing the dtmf? |
| 16:28.57 | lal00 | the client is a regular phone, it dials an 800 number and it gets redirect via sip to my server |
| 16:29.56 | pigpen | so you are saying that the remote caller's dtmf is not being recognized when they, lets say, press 1 on a menu? |
| 16:30.07 | saxa | [TK]D-Fender: i will step later into it drastically, by reconfiguring the dahdi to the minimum necessary to work. So far I have added callprogress=no to chan 1 but sincerly speaking, nothing changes, i have restarted both, dahdi ans * . |
| 16:30.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:31.09 | lal00 | pigpen: if they press 1234567890, for example, i would only get 9 digits. This happens in 1 out of 4 calls. |
| 16:31.22 | lal00 | (aprox.) |
| 16:31.23 | emate | _Corey_: yes, i want to set variable that will be available in all actions (forward call to another person in office etc.) so i can identify all actions taken for this call. |
| 16:32.09 | pigpen | lal00, so what is your sip gateway equipment? Audiocodes for example has a mess of things that can affect dtmf recognition |
| 16:32.40 | lal00 | pigpen: I need to ask that one. |
| 16:33.14 | emate | _Corey_: in cdr i have uniqueid, but it differs when call is forwarded to another person in office. I don't know how to group this two actions. |
| 16:33.21 | lal00 | meanwhile, is there anything I could check in the logs or any settings that can help me find if something is clearly wrong on my side? |
| 16:33.28 | [TK]D-Fender | mentax verify that you have a proper indications.conf in place |
| 16:34.38 | pigpen | lal00, without knowing the parts involved and where the possibilities may lie, it would be a waste of time. You need manufacturers, models, firmware, settings, configurations?? |
| 16:36.20 | pigpen | [TK]D-Fender, do you know if the "Set(CHANNEL(buffers)=?12,half?" in the dial plan can be applied to a sip t.38 to sip t.38 vs. a dahdi to sip t.38? |
| 16:36.45 | [TK]D-Fender | pigpen, Never touched any part of that |
| 16:36.50 | pigpen | ie: it is documented as "DAHDI Buffer Policy Implementation" |
| 16:37.05 | pigpen | [TK]D-Fender, thanks. I hope you feel bad for me. ;-) |
| 16:37.27 | pigpen | when in doubt, try it and find out. |
| 16:37.43 | ndespres | any NYC asterisk pros in here? |
| 16:39.26 | pigpen | ? please, not all at once. |
| 16:41.21 | pigpen | ndespres, what do you need. I cannot speak for the rest, but the bulk of asterisk management we do is across the country. |
| 16:41.30 | pigpen | many here will be happy to help. |
| 16:41.58 | citywok | emate: check out astdb and see if it does what you need |
| 16:42.15 | Kroms | has anyone ever worked with LIME SIP provider? |
| 16:43.21 | _Corey_ | emate: unique id may not be what you need... the asterisk variable handling and astdb stuff is explained pretty well in the Asterisk book |
| 16:43.31 | _Corey_ | ~thebook |
| 16:43.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:43.36 | _Corey_ | emate: ^^^^ |
| 16:43.39 | ndespres | pigpen, i work for a small business IT consulting firm, we just acquired another company that had deployed TrixBox installations at 5 small offices around here.. |
| 16:43.40 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:44.07 | pigpen | you may want to check out the trixbox irc channel. |
| 16:44.17 | pigpen | many here find it evil. (as do I) |
| 16:44.21 | ndespres | and i simply don't have the bandwidth or knowledge to manage this stuff wihtout impacting my clients. I was hoping to get someone else to manage this relationship. |
| 16:44.54 | ndespres | Essentially outsourcing this work/relationship to someone more knowledgeable than I. |
| 16:45.05 | _Corey_ | ndespres: Contact me privately if you want... I'm in NYC on Thursday |
| 16:45.06 | citywok | ndespres: it would be good to say where you are located so somebody in your area can chime in |
| 16:45.13 | pigpen | yeah, it is just a pain in their logic. this channel is more for the hard core, design your own. |
| 16:45.21 | citywok | ndespres: sorry i just saw it a few lines up, my fault |
| 16:45.28 | ndespres | yes, i'm quickly learning that trixbox is a PITA |
| 16:45.37 | pigpen | ndespres, there you go?_Corey_ knows what he is doing. |
| 16:45.55 | _Corey_ | (sometimes) ;) |
| 16:45.59 | [TK]D-Fender | Trixabox has been dead a very long time |
| 16:46.26 | citywok | [TK]D-Fender: aren't you always n the trixbox channel? |
| 16:46.28 | pigpen | _Corey_, hey, go with it. Nobody need to know the truth. |
| 16:46.46 | [TK]D-Fender | citywok, Nope |
| 16:46.50 | citywok | oh maybe that was freepbx i saw you in |
| 16:47.27 | pigpen | Fyi, setting CHANNEL(buffers)= on a sip T.38 rather than a DAHDI does not work, fyi. |
| 16:47.51 | ndespres | thanks for your help, everyone |
| 16:48.05 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 16:55.24 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 16:55.26 | DelphiWorld | hey all |
| 16:55.37 | DelphiWorld | please can someone tel me how to play an anouncemant in early media mode? |
| 16:55.54 | [TK]D-Fender | DelphiWorld, "core show application playback" |
| 16:56.05 | DelphiWorld | [TK]D-Fender: please... show me how ;) |
| 16:56.11 | [TK]D-Fender | DelphiWorld, read the instructions |
| 16:56.33 | DelphiWorld | [TK]D-Fender: i don't have a astbox runing right now so i'm just wondring:P |
| 16:56.42 | DelphiWorld | any idea pabelanger |
| 16:56.48 | [TK]D-Fender | DelphiWorld, Keep wondering then :0 |
| 16:57.36 | [TK]D-Fender | DelphiWorld, the apps instructions are available all over the palce as it is |
| 16:57.39 | [TK]D-Fender | place* |
| 16:58.58 | DelphiWorld | [TK]D-Fender: you back to your idiotic aguin ? |
| 16:59.14 | DelphiWorld | [TK]D-Fender: want to be banned aguin so ? |
| 17:00.52 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:01.05 | [TK]D-Fender | DelphiWorld, Me idiotic? Never. |
| 17:01.21 | DelphiWorld | sory [TK]D-Fender but you look realy unsane |
| 17:01.37 | DelphiWorld | i must ask Qwell to react:) |
| 17:01.59 | [TK]D-Fender | DelphiWorld, Coming from someone who is supposedly legally blind... "look" isn't a word you should be throwing around ..... |
| 17:02.19 | DelphiWorld | ok so [TK]D-Fender looking not only through eyes. do you know that ? |
| 17:02.49 | puzzled | DelphiWorld: you could have done a quick search on the asterisk wiki and find your answer faster than asking here. hint: use option noanswer |
| 17:02.57 | [TK]D-Fender | DelphiWorld, Stop being a lazy troll and go read something for yourself :p |
| 17:03.08 | DelphiWorld | puzzled: that was i'm doing, dear |
| 17:03.11 | puzzled | yes that's another way of putting it :) |
| 17:03.12 | [TK]D-Fender | puzzled, DO NOT FEED! |
| 17:03.47 | *** join/#asterisk Linuturk_ (~linuturk@unaffiliated/linuturk) |
| 17:03.52 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 17:03.54 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-kkbjqtamnlqiyeaz) |
| 17:03.59 | DelphiWorld | [TK]D-Fender shut up cause puzzled have a nice way communicating thant you :P |
| 17:04.26 | puzzled | DelphiWorld: no I agree with [TK]D-Fender that you are being lazy and should make an effort to find your answer before asking here |
| 17:04.45 | DelphiWorld | puzzled: yes, but you did it in the right way |
| 17:05.07 | puzzled | so instead of causing silly discussions in this channel, next time just Google... |
| 17:07.04 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 17:08.01 | DelphiWorld | puzzled: thank you |
| 17:08.06 | DelphiWorld | beat [TK]D-Fender ass. |
| 17:08.14 | DelphiWorld | exten => s,1,Progress exten => s,n,Playback(followme/pls-hold-while-try) |
| 17:08.14 | DelphiWorld | d |
| 17:08.23 | DelphiWorld | :D |
| 17:19.19 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 17:21.31 | DelphiWorld | exten => s,n,Playback(followme/pls-hold-while-try|noanswer) |
| 17:23.37 | *** join/#asterisk sysdef (sysdef@debiancenter/founder.developer/pdpc.professional.sysdef) |
| 17:24.54 | p3nguin | Only if you use asterisk 1.2 |
| 17:25.12 | p3nguin | We stopped using the pipe line after that. |
| 17:25.13 | [TK]D-Fender | He doesn't even use asterisk... |
| 17:25.34 | p3nguin | That's annoying. |
| 17:27.50 | ShaunR | is there a way to create entrys in the queue_log? |
| 17:28.49 | p3nguin | Speaking of annoying, how are people in India with India phone numbers able to accidentally call my US toll-free number? Wouldn't they have to dial the country code of 1 to reach the US? |
| 17:29.41 | p3nguin | It's a sudden problem for a bunch of people, it seems. |
| 17:29.59 | p3nguin | I google the numbers and they show up in indiatrace. |
| 17:30.07 | p3nguin | But so does my toll-free number! |
| 17:30.15 | puzzled | heh |
| 17:30.35 | WIMPy | hi puzzled |
| 17:30.41 | puzzled | hey WIMPy |
| 17:31.10 | WIMPy | puzzled: Just read your mail. I also asked you here if you had two versions installed, but I guess you didn't see that any more. |
| 17:31.35 | puzzled | p3nguin: maybe there's some Indian telco that allows calling of TF numbers |
| 17:32.10 | puzzled | WIMPy: the error was all mine. I needed to specify the actual kernel module name (without .ko) in the search override file in /etc/depmod.d/ |
| 17:33.17 | puzzled | once I corrected that the problem went away. it took me reading the depmod.conf manpage a few times before brain started working. learn something every day. slowly :) |
| 17:33.22 | WIMPy | Yes, I used to be under the impression that extra takes preference for quite some time, but that's not the case. |
| 17:33.32 | p3nguin | 8136850xxx appears to be a valid NANP number, but it's from India. I'm suddenly getting a shitload of calls from doctors' numbers which are in India. What's up? |
| 17:33.36 | WIMPy | :-) |
| 17:34.04 | puzzled | turn on the screaming monkey sound :) |
| 17:34.19 | puzzled | scares the hell out of unsuspecting people |
| 17:34.39 | WIMPy | p3nguin: But 81 is Japan. |
| 17:35.00 | WIMPy | How does India fit in there? |
| 17:35.05 | WIMPy | (Or the number) |
| 17:35.23 | puzzled | creative routing it seems |
| 17:38.50 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 17:38.59 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 17:41.12 | *** join/#asterisk molnarp (~molnarp@nx7400.ohsh.u-szeged.hu) |
| 17:41.53 | WIMPy | waits for feedback for his MSN matching code. |
| 17:44.38 | *** part/#asterisk pietro (~pietro@88-149-226-83.dynamic.ngi.it) |
| 17:45.27 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 17:51.28 | *** join/#asterisk fireman_biff (~biff@69.73.217.114) |
| 17:53.02 | DelphiWorld | coppice: go out ;) |
| 17:57.07 | *** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 18:15.39 | *** join/#asterisk Wolfeyes (~Wolfeyes@41-133-96-190.dsl.mweb.co.za) |
| 18:24.42 | *** part/#asterisk Wolfeyes (~Wolfeyes@41-133-96-190.dsl.mweb.co.za) |
| 18:32.49 | *** join/#asterisk koffel (koffel@173-167-212-105-ip-static.hfc.comcastbusiness.net) |
| 18:33.02 | koffel | hey all |
| 18:33.14 | *** join/#asterisk mazpe (~lesterm@li315-205.members.linode.com) |
| 18:33.27 | koffel | can i use asterisk and my cell phone 3g network? |
| 18:33.46 | mazpe | any recommendations on how to convert a g729 file to ulaw? |
| 18:34.27 | WIMPy | 'file convert' ... |
| 18:35.34 | koffel | i guess not? |
| 18:35.35 | p3nguin | wimpy: 10 digits. Does Japan have 10 digit numbers starting with 81? |
| 18:36.12 | WIMPy | I have NFI how long japanese numbers are. |
| 18:36.44 | WIMPy | koffel: Sure. Do you also want to link them together? |
| 18:37.08 | koffel | yes when i am out on the road |
| 18:37.19 | koffel | i wanta be able to recieve a call |
| 18:37.19 | mazpe | file.c:186 ast_writestream: Unable to translate to format pcm, source format g729 |
| 18:37.48 | WIMPy | koffel: Where does Asterisk come in there? |
| 18:37.56 | WIMPy | mazpe: Do you have a licence? |
| 18:38.13 | mazpe | i have g729 |
| 18:38.13 | koffel | i have asterisk as my home number |
| 18:38.16 | mazpe | license. |
| 18:43.15 | Qwell | mazpe: Do you have the codec installed? O.o |
| 18:43.25 | Qwell | Do you have enough licenses? Any active channels when you tried to do it? |
| 18:43.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:43.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:50.09 | p3nguin | I think I'm about to get to the bottom of all the calls from random doctors. |
| 18:50.47 | Qwell | is it Lupus? |
| 18:51.10 | p3nguin | Some discount prescription company just fax spammed a bunch of doctors' offices and added "To opt out of fax messages, call ..." (they used my number). |
| 18:51.18 | Qwell | awesome |
| 18:51.50 | p3nguin | So they are all calling to opt out, I reckon. I personally only talked to one caller. |
| 18:52.05 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 18:52.10 | p3nguin | She didn't say anything about the opt out thing, though. |
| 18:52.27 | Qwell | p3nguin: You could throw a message on the IVR about it. "If you are a doctors office, press 9" |
| 18:52.32 | p3nguin | So I ended up calling the most persistent caller, which turned out to be a fax number. |
| 18:53.14 | p3nguin | I googled their CID name and got their regular number, and called it. That's where I found the girl that was able to tell me about the opt out crap. |
| 18:58.06 | p3nguin | That sounds like it could work. I think I'm going to have to do that. |
| 18:58.30 | p3nguin | So I'm on the line with the source of the fax spam... |
| 18:58.36 | paulc | Haha that's awesome.. I get occasional calls for a medical supply company.. some literature somewhere has their tollfree number off by 1 digit, compared to mine.. fun fun fnu |
| 18:59.51 | *** join/#asterisk dxd828 (~dxd828@88-109-125-30.dynamic.dsl.as9105.com) |
| 19:00.03 | *** part/#asterisk dxd828 (~dxd828@88-109-125-30.dynamic.dsl.as9105.com) |
| 19:00.24 | p3nguin | This person is not happy with my call. I told her I need to talk to the person responsible for the fax, and she put me on hold. She came back and asked me where I'm calling from. I advised her that isn't relevant, that they are the ones sending the fax out, but she put me back on hold before I got to finish my sentence. |
| 19:02.04 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 19:02.35 | mazpe | Qwell: g729 is installed and currently been utilized. |
| 19:02.35 | _Corey_ | p3nguin: I'd kindly offer them a free IVR to handle their calls (unless they wanted to pay me for an alternate solution): "Press 1 to lodge a complaint with the FTC" |
| 19:02.49 | *** join/#asterisk chasing`Sol (~cS@197.132.183.5) |
| 19:03.00 | p3nguin | She told me she knew that an email was sent out, but knows nothing about a fax. |
| 19:03.18 | p3nguin | If they don't agree to resolve the issue, I'll be calling the FTC and FCC myself. |
| 19:03.36 | *** join/#asterisk fireman_biff (~biff@69.73.217.114) |
| 19:04.12 | [TK]D-Fender | mazpe, Go prove you can convert to something else |
| 19:04.40 | _Corey_ | p3nguin: I'd play everyone who calls and chooses the option a helpful recording identifying the company and transfer them to the FTC |
| 19:04.42 | _Corey_ | ;) |
| 19:04.54 | mazpe | [TK]D-Fender: well, i was trying to go from g729 to ulaw |
| 19:05.16 | [TK]D-Fender | mazpe, show us |
| 19:07.12 | p3nguin | Yay, a supervisor is cooperating. |
| 19:09.51 | *** part/#asterisk fireman_biff (~biff@69.73.217.114) |
| 19:13.08 | p3nguin | Nice. She found the fax document, and confirmed that it does have my number on it. |
| 19:14.34 | Qwell | call the FCC |
| 19:14.43 | Qwell | get them slammed with fines |
| 19:19.39 | p3nguin | The supervisor was cooperative. She indicated that she is going to try to find out how this happened. She's also escalating it and someone is supposed to fix the number and also let me know how it happened. If I find out that they are not fixing it, then I will call the FCC. |
| 19:20.27 | _Corey_ | Gah, unless they paid me off for my annoyance I'd be totally filing complaints |
| 19:23.06 | p3nguin | I like calls from doctors' offices, but not so much that I want them all calling at once ringing the phones like mad. |
| 19:23.16 | p3nguin | I even got a fax call to my cell phone because of it. |
| 19:25.11 | mazpe | whats that command again to show the g729 licenses? |
| 19:25.15 | bbourdage | Does anyone have any thoughts on this ?. We have 2 asterisk boxes using cdr_odbc,, writing to a mysql box, it was working well and then we started getting the following errors. The data is still written to the database, we have checke dthe users and the database and it looks good, and the mysql log shows that it was written and does not return the error to asterisk ?[2012-02-21 13:16:55] ERROR[13572]: cdr_sqlite.c:183 |
| 19:25.25 | p3nguin | g729 show licences, perhaps. |
| 19:25.49 | p3nguin | Of course you have to spell it right. |
| 19:26.06 | p3nguin | lic TABKEY |
| 19:26.41 | Qwell | bbourdage: What is the full error? |
| 19:26.58 | bbourdage | ] ERROR[14099]: cdr_sqlite.c:183 sqlite_log: cdr_sqlite: attempt to write a readonly database |
| 19:27.19 | bbourdage | That is all I get, I have a high loggin glevel also |
| 19:27.26 | Qwell | if you aren't using cdr_sqlite, disable it in the config |
| 19:29.26 | bbourdage | Which config file ?, I do not have it in the asterisk.conf, or the cdr_adaptaptive_odbc |
| 19:29.54 | Qwell | cdr_sqlite.. |
| 19:30.18 | bbourdage | I do not have that text file in the asterisk directory ? |
| 19:30.41 | Qwell | ls -l /etc/asterisk/cdr*.conf |
| 19:30.47 | Qwell | What do you see? |
| 19:31.27 | bbourdage | /etc/asterisk# grep -i sql * |
| 19:31.27 | bbourdage | cdr_adaptive_odbc.conf:;connection=mysql1 |
| 19:31.27 | bbourdage | cdr_adaptive_odbc.conf:;connection=sqlserver |
| 19:31.38 | Qwell | I didn't say to grep, did I? |
| 19:32.01 | bbourdage | I had done that first, doing youir command now |
| 19:32.27 | bbourdage | /etc/asterisk# ls -l /etc/asterisk/cdr*.conf |
| 19:32.27 | bbourdage | -rw-r--r-- 1 root root 2619 Jan 17 22:52 /etc/asterisk/cdr_adaptive_odbc.conf |
| 19:33.08 | bbourdage | I know, root not good, I am changing |
| 19:34.52 | [TK]D-Fender | bbourdage, ls -la /usr/lib/asterisk/modules |
| 19:35.09 | [TK]D-Fender | bbourdage, then "noload => " the ones you don't need in modules.conf |
| 19:36.18 | bbourdage | That is it !!!!, what is the best CDR to use, the cdr adaptive does not record time and date of call ! |
| 19:36.20 | mazpe | [TK]D-Fender: ok... so g729 show licenses: 0/0 encoders/decoders of 5 licensed channels are currently in use |
| 19:39.05 | Qwell | bbourdage: umm, it'll record whatever you tell it to record. hence the "adaptive" |
| 19:40.09 | bbourdage | I was looking for the variable to use, is the best way to look at the code, or does anyone have a link that lists any of the variables you can use ?. Thanks for everyones help, in 4 minutes, you solved, what I put 2 hours into, thinking it was the SQL side. |
| 19:41.04 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 19:42.19 | [TK]D-Fender | bbourdage, there is a compile option to have the datetime in there IIRC |
| 19:42.34 | bbourdage | Thanks |
| 19:44.27 | slingr | I have a bunch of entries starting last night in my CDR http://pastebin.ca/2120510 |
| 19:44.42 | slingr | is this someone trying to get into my box on an anonymous extension? |
| 19:45.32 | WIMPy | Extensions are in your dialplan and never anonymous. |
| 19:46.16 | slingr | understood. any idea what those could be. IP says florida and i'm in toronto |
| 19:46.38 | slingr | and it says ANSWERED and I wasn't up at 5am to answer 20+ 13s calls |
| 19:46.58 | WIMPy | So what does your s extension do? |
| 19:47.10 | WIMPy | Assuming the s there is the destination of the call. |
| 19:48.27 | slingr | as far as I know, there is only the 1 extension I have, 101 , at least thats what freepbx says. |
| 19:48.55 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 19:49.01 | slingr | iirc, asterisk uses extensions.conf? |
| 19:49.57 | slingr | found s |
| 19:51.43 | slingr | maybe not |
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| 20:06.19 | bbourdage | Does anyone have any recomendations for a GUI client at the desktop, commercial or free . |
| 20:06.53 | *** join/#asterisk cmendes0101 (~nn@pool-173-58-92-161.lsanca.fios.verizon.net) |
| 20:07.06 | _Corey_ | bbourdage: You should probably take a look at Switchvox if you haven't already |
| 20:07.34 | bbourdage | I never realized that would work with plain Asterisk, great idea. |
| 20:07.56 | _Corey_ | bbourdage: Who said anything about it working with plain Asterisk...? ;) |
| 20:08.29 | _Corey_ | Switchvox has Asterisk "under the hood" but it's meant to be a turn-key sort of thing... |
| 20:08.41 | bbourdage | Ok, I should have defined that in my request, sorry. Will it work ? |
| 20:09.52 | _Corey_ | depends on what you're asking |
| 20:09.55 | _Corey_ | if your question is "Will Switchvox drop onto my existing Asterisk implementation and give my end-users a GUI?" then no. |
| 20:14.22 | bbourdage | Is it a lot of work to make it work, or it will not happen at all ?, and what other options are out there ? |
| 20:14.57 | _Corey_ | bbourdage: You need to be more clear about what you need in an end-user GUI, what you have in place, what you're willing/able to change, etc. |
| 20:15.38 | bbourdage | Currently, call control, voice mail, maybe status conditions ?, to start |
| 20:16.38 | [TK]D-Fender | bbourdage, Thre is nothing "drop-in" worthy out there. |
| 20:16.52 | [TK]D-Fender | bbourdage, Everything is "Souls sold to the lowest bidder" |
| 20:17.54 | bbourdage | Sounds like an oppurtunity for a commercial product |
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| 22:42.07 | ChannelZ | Wow, just got some SIP trolling from someplace other than China for once. |
| 22:43.13 | molnarp | hi, is anyone there, who is familiar with the Digium T410P card and its VPMADT032 echo canceller module? It seems that I have some firmware loading issues |
| 22:43.53 | Qwell | molnarp: call up Digium support. That's why they're there. |
| 22:44.33 | molnarp | I did. Their support is worthless. |
| 22:45.52 | molnarp | I'm using debian stable release, upgrade to unstable/testing is not an option, installing from source also not an option. They say these versions are unsupported. Nothing more. |
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| 22:46.11 | Qwell | well there you go then |
| 22:46.27 | Qwell | If the drivers in that version don't support that card, there's nothing you can do. |
| 22:47.12 | molnarp | but they do, at the time they were released, this card was in production for a long time |
| 22:48.14 | molnarp | the first thing I'd like to know, that what happens if i load the wrong firmware to the VPMADT032, or load no firmware at all |
| 22:48.15 | Qwell | TDM410 is a new model, that supports the echo can module. You're thinking TDM400p |
| 22:49.35 | molnarp | no, i think of TDM410P with the VPMADT032 echo can module |
| 22:50.03 | molnarp | the question is, would the wctdm24xxp module recognize the echo can module, if there's no firmware, or wrong firmware in it? |
| 22:50.24 | *** join/#asterisk abesamthomas (~abesamtho@61.17.32.29) |
| 22:50.37 | Qwell | What version of Debian? (stable doesn't say much) |
| 22:51.07 | *** join/#asterisk abesamthomas (~abesamtho@61.17.32.29) |
| 22:51.07 | molnarp | does the echo can module has some sort of "basic" functionality without firmware loaded? or it is completely dumb without it? |
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| 22:51.17 | molnarp | debian squeeze (6.0) |
| 22:53.13 | molnarp | i have compiled the dahdi module using module-assistant, during build, it downloaded the firmware from Digium, we analyzed the build log with tzafrir, he said the firmware is correctly "compiled" into it |
| 22:53.39 | saxa | [TK]D-Fender, kaldemar, ok, I got it working, I redid a chan_dahdi.conf file with as minimum options as possible, and it started working. Thanks guys for your time and help. |
| 22:53.41 | molnarp | but, the echo can module is still: VPM100: Not Present |
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| 23:05.44 | saxa | [TK]D-Fender, kaldemar , here is the paste of the chan_dahdi.conf and dahdi-channels.conf which are working. Seems that some options which i had before (the ones commented out are all the options I had) were conflicting. I suspect that the answeronpolarity=yes and hanguponpolarity=yes where the ones. Although I do not know why I enabled those :). |
| 23:05.53 | saxa | http://pastebin.com/KSrSj7j5 |
| 23:06.04 | *** join/#asterisk funkylonehat (~funkylone@125-236-222-73.adsl.xtra.co.nz) |
| 23:07.48 | saxa | now I have only one more problem, and is why my phone won't register behind a nat. But this is for tomorrow :) good stay in the chan to all. |
| 23:08.09 | funkylonehat | ty saxa. all the best. :) |
| 23:11.06 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 23:11.13 | ruben23 | hi guys |
| 23:12.37 | ruben23 | any help on this error---> i uploaded a audio fromat .wav adn when i tried to play it on asterisk logs says ----> WARNING[4136]: file.c:653 ast_openstream_full: File custom/TestWelcome does not exist in any format |
| 23:15.07 | ruben23 | guys any idea how to correct this..? |
| 23:16.16 | [TK]D-Fender | provide the file it's looking for |
| 23:23.24 | p3nguin | Too easy. Got anything more difficult to try? |
| 23:24.49 | ruben23 | <PROTECTED> |
| 23:25.21 | [TK]D-Fender | ruben23: prove that the right file is in the right place |
| 23:25.49 | p3nguin | Show me the output from: namei -mo /var/lib/asterisk/sounds/custom/TestWelcome* |
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| 23:26.35 | p3nguin | and/or: namei -mo /var/lib/asterisk/sounds/en/custom/TestWelcome* |
| 23:26.53 | p3nguin | pastebin it. |
| 23:28.42 | ruben23 | <PROTECTED> |
| 23:29.59 | [TK]D-Fender | ruben23: well you were the one who told it there was |
| 23:30.15 | [TK]D-Fender | ruben23: File custom/TestWelcome <---------- |
| 23:30.17 | puzzled | p3nguin: that's a cool command. didn't know about that one |
| 23:30.46 | p3nguin | I like it, too. |
| 23:34.36 | ruben23 | sorry about that |
| 23:35.25 | ruben23 | i pasted the log error form the google search i made, this is my actual erro-n directory specified--> [Feb 21 15:33:40] WARNING[14405]: file.c:664 ast_openstream_full: File Fidelity_voice does not exist in any format |
| 23:35.35 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 23:35.35 | *** mode/#asterisk [+o file] by ChanServ |
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| 23:39.15 | p3nguin | namei -mo /var/lib/asterisk/sounds/Fidelity_voice /var/lib/asterisk/sounds/*/Fidelity_voice |
| 23:39.21 | p3nguin | crap |
| 23:39.29 | p3nguin | namei -mo /var/lib/asterisk/sounds/Fidelity_voice.* /var/lib/asterisk/sounds/*/Fidelity_voice.* |
| 23:39.39 | p3nguin | Disregard the first one before "crap" |
| 23:39.56 | p3nguin | Pastebin the results. |
| 23:46.43 | ruben23 | i find it on ---> root@vicidial:/var/www/4n3zrqn0ydxdc5gchkxmc0dphj6yp9# ls Fidelity_voice.wav |
| 23:47.00 | ruben23 | /var/lib/asterisk/sounds/Fidelity_voice.* /var/lib/asterisk/sounds/*/Fidelity_voice.* <---no result |
| 23:47.20 | p3nguin | There's yer problem. |
| 23:49.00 | p3nguin | Unless you have configured asterisk to look in /var/www/4n3zrqn0ydxdc5gchkxmc0dphj6yp9 for sound files, or unless asterisk is pure magic, no one would expect that to work. |
| 23:50.24 | ruben23 | <PROTECTED> |
| 23:50.57 | p3nguin | Show me the output of: file Fidelity_voice.wav |
| 23:53.05 | ruben23 | <PROTECTED> |
| 23:53.43 | p3nguin | Example: file /var/lib/asterisk/sounds/local/developer.wav |
| 23:53.43 | p3nguin | /var/lib/asterisk/sounds/local/developer.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 23:54.40 | ruben23 | root@vicidial:/var/www/4n3zrqn0ydxdc5gchkxmc0dphj6yp9# file Fidelity_voice.wav ---> Fidelity_voice.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 23:54.41 | ruben23 | <PROTECTED> |
| 23:55.18 | p3nguin | Good. Copy Fidelity_voice.wav to the relevant sounds directory for asterisk and make sure asterisk has permission to read it. |
| 23:57.35 | ruben23 | /var/www/4n3zrqn0ydxdc5gchkxmc0dphj6yp9# cp Fidelity_voice.wav /var/lib/asterisk/sounds <---done |
| 23:58.47 | ruben23 | whats next..? |
| 23:59.01 | p3nguin | Try your call again to see if the sound plays. |