| 00:03.25 | volga629 | because Object class violation what this error mean ? |
| 00:03.29 | volga629 | res ldap |
| 00:17.20 | *** join/#asterisk greenwolf (~penguin@cpe-72-228-137-112.buffalo.res.rr.com) |
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| 01:22.22 | Ice_Strike | [May 26 03:12:23] WARNING[3217]: chan_sip.c:3666 retrans_pkt: Retransmission timeout reached on transmission YzllNmNmMWRlNWNiNjk4MzQxZDdkMDkwZGZkZmNmMzA. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 01:22.31 | Ice_Strike | whats causing this? |
| 01:23.00 | Ice_Strike | in sip debug on: |
| 01:23.10 | Ice_Strike | Retransmitting #4 (NAT) to 94.192.xx.xx:52565: |
| 01:23.26 | jpsharp | packet loss |
| 01:23.29 | Ice_Strike | is it because of this? |
| 01:30.55 | WIMPy | Networking/firewall issue or the peer has died. |
| 01:32.49 | Ice_Strike | http://pastebin.com/ZLNNAvUk |
| 01:33.01 | Ice_Strike | I think this is an issue |
| 01:33.02 | Ice_Strike | Retransmitting #4 (NAT) to 94.192.11x.xxx:52565: |
| 01:33.11 | Ice_Strike | 94.192.11x.xxx:52565 is my softphone |
| 01:38.07 | Ice_Strike | any idea? |
| 01:38.25 | Ice_Strike | On the firewall, I have 5060 - 5082 PORT |
| 01:38.31 | Ice_Strike | and 8000 - 8100 |
| 01:39.48 | WIMPy | That obviousely doesn't include 52565. |
| 01:40.24 | WIMPy | However usually connection tracking should take care of that anyway. |
| 01:45.10 | Ice_Strike | I am accessing asterisk remotely via public IP |
| 01:45.15 | Ice_Strike | dont seem to work properly |
| 01:45.39 | Ice_Strike | It work when I test on the LAN |
| 01:46.44 | volga629 | HAS NOT USED (OR SWITCHED TO) TLS in favor of 'UDP' (but this was allowed in sip.conf)! |
| 01:46.50 | volga629 | what is mean ? |
| 01:53.58 | Ice_Strike | I give up, I will do it on LAN. |
| 02:09.40 | volga629 | How I cant set priority for extconfig ldap |
| 02:11.24 | volga629 | can |
| 02:39.56 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:39.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:42.15 | volga629 | why res_config_ldap.c: Couldn't modify it trying modify entries in ldap server ? |
| 02:45.34 | volga629 | ERROR[9420]: res_config_ldap.c:1328 update_ldap: |
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| 03:51.50 | soapee45 | anyone around this evening? |
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| 04:31.47 | SeRi | p3nguin: you around? |
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| 04:53.41 | din3sh | mrning all |
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| 07:27.27 | oyugik | Hey goodmorning from nairobi |
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| 08:44.02 | oyugik | anybody know hot to configure analogue phones to use asterisk extensions |
| 08:46.02 | atan | So I went from 10.3 to 10.4 and one of my cisco phones won't connect properly using realtime =\ color me confused nothing else really changed O_O |
| 08:46.41 | atan | When trying to dial outbound it goes to the default context, and device itself doesn't show as being registered in the database... but sip show peers shows a cached one somehow |
| 08:51.27 | *** join/#asterisk twanny796 (~twanny@46.11.8.201) |
| 08:53.45 | *** join/#asterisk bintut (~bintut@cm102.sigma15.maxonline.com.sg) |
| 08:53.56 | bintut | waves |
| 08:55.28 | bintut | i am running asterisk-1.8.11 from the digium debian binary packages. i loaded in my modules.conf the res_convert.so. i'm wondering how to properly use this module in order for me to convert my audio files into g729 format. or, any recommended tool that i can use to convert audio files into g729 format? |
| 08:56.12 | din3sh | interesting stuff --> http://www.eyepea.eu/case-studies/migrating-from-cisco-call-manager-to-asterisk |
| 08:56.51 | din3sh | atan: which cisco phone? |
| 08:56.57 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 08:58.26 | din3sh | atan2: which cisco phone? |
| 08:58.45 | bintut | anyone? |
| 08:59.56 | atan2 | din3sh, 7960 :) |
| 09:00.31 | atan2 | din3sh, going back to 10.3 just to see if it is, 100% this or something else. There was a server move in there but it was setup the same way, I swear... using the same database, same config files, same everything. |
| 09:08.44 | din3sh | which firmware on the phone? |
| 09:08.49 | din3sh | 8.x or 9.x? |
| 09:08.54 | atan2 | Well drat. Something else must have changed 10.3 is fooy. |
| 09:08.58 | atan2 | 8.something I do believe |
| 09:10.28 | din3sh | i updated firmware to 9 |
| 09:10.43 | din3sh | had real pain in the ass to get it working with 1.8 |
| 09:11.27 | din3sh | bintut: what files do you want to convert? |
| 09:12.11 | bintut | din3sh: audio files for ivr. i have g729a licenses from digium. |
| 09:13.21 | bintut | this is for my home pbx and i have 2 x g729a licenses |
| 09:15.24 | bintut | i'm not sure how am i going to convert my audio files. i have 3 different file extensions: .raw, .pcm and .gsm |
| 09:15.28 | bintut | please advice. |
| 09:16.06 | din3sh | you've tried the file convert command? |
| 09:17.46 | *** join/#asterisk angryuser_laptop (~angryuser@172.237.146.195.dynamic.adsl.abo.nordnet.fr) |
| 09:18.13 | bintut | din3sh: how is it? |
| 09:19.45 | din3sh | file convert /var/lib/asterisk/sounds/custom/file.wav /var/lib/asterisk/sounds/custom/file.g729 |
| 09:21.48 | atan2 | Well blarg. I'm confused. sip show peers will show the phone. But, no outbound for me, not inbound... but it's there, sort of, kind of, maybe. I best review that realtime tutorial something must have been missed. |
| 09:22.05 | bintut | din3sh: i just did similar command. kindly check it here => http://www.pastie.org/3970425 |
| 09:26.26 | atan2 | Anyone recall the link to that Asterisk realtime quickstart guide that is on the wiki? |
| 09:30.18 | din3sh | bintut: Unable to open input file: /tmp/menu.raw |
| 09:30.36 | din3sh | do you have proper access to the folder? |
| 09:30.49 | din3sh | read/write |
| 09:31.35 | din3sh | atan2: https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
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| 09:35.09 | *** part/#asterisk oyugik (~oyugik@41.84.143.52) |
| 09:37.04 | atan2 | Not sure where but somewhere I think my odbc connection is pooched... "[ISQL]ERROR: Could not SQLConnect" |
| 09:40.16 | din3sh | credentials? |
| 09:40.57 | atan2 | I'll vrevert to the old password and see what happens |
| 09:45.01 | atan2 | Well thereis something going on there I'm just not sure what yet. Hmm. res_odbc.c:654 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]... |
| 09:52.53 | *** join/#asterisk shadebob (~shadebob@adsl196-45-235-206-196.adsl196-8.iam.net.ma) |
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| 10:41.36 | bintut | din3sh: sorry, i was out. yes, the /tmp folder has a 1777 permission |
| 10:50.18 | bintut | how to test an audio file was converted properly to g729? the audio file was converted from from raw or wav format using asterisk's res_convert module. |
| 11:04.28 | bintut | hello..? |
| 11:04.31 | bintut | my asterisk v1.8.11 home pbx does not transcode and does not handle g729 codec despite of having 2 x g729 codec license bought from digium. when i place a sip call using g729 as the preferred codec, it does not work as what i expected. first. when it reaches the ivr to play the menu.g729 file, it plays so fast that i don't understand it says. second, when it dials to the destination channel dahdi/4-1, i am getting a warning message that the |
| 11:04.32 | bintut | analog phone does not ring. |
| 11:04.47 | bintut | [May 26 18:52:16] WARNING[25876]: channel.c:4913 ast_write: Codec mismatch on channel DAHDI/4-1 setting write format to g729 from ulaw native formats 0x4 (ulaw) |
| 11:10.54 | *** join/#asterisk shadebob (~shadebob@adsl196-45-235-206-196.adsl196-8.iam.net.ma) |
| 11:12.06 | din3sh | analog phone? |
| 11:12.11 | din3sh | via fxs card? |
| 11:16.01 | bintut | yes, via fxs |
| 11:29.46 | din3sh | the phone does ring with ulaw, right? |
| 11:39.40 | bintut | yes, it is |
| 11:40.08 | bintut | actually, no problem earlier before i added g729 in my sip.conf |
| 11:43.12 | din3sh | have u loaded the g729 properly? |
| 11:43.19 | bintut | din3sh: kindly check this => http://www.pastie.org/3970890 |
| 11:46.35 | din3sh | sip.conf setting for that extension? |
| 11:49.34 | bintut | din3sh: the only added info in my sip.conf is g729 from: |
| 11:49.45 | bintut | allow=ulaw,alaw,gsm |
| 11:49.47 | bintut | to |
| 11:49.57 | bintut | allow=g729,ulaw,alaw,gsm |
| 11:51.49 | bintut | in the [general] section |
| 11:55.41 | bintut | any idea? please advice. |
| 11:56.20 | bintut | what am i missing here? did i install the g729 module wrongly or configure it wrongly? |
| 11:58.55 | din3sh | show your sip.conf |
| 12:00.59 | bintut | or, is it the codec_g729a-1.8.4_3.1.5-opteron_sse3_32.tar.gz is not correct for my intel D945GCLF2 motherboard with an on-board intel atom cpu 330 @ 1.60GHz? |
| 12:02.05 | din3sh | might be |
| 12:02.16 | din3sh | show your sip.conf |
| 12:03.07 | din3sh | have you set disallow=all before allow=g729...? |
| 12:07.02 | bintut | din3sh: kindly refresh http://www.pastie.org/3970890 |
| 12:07.10 | bintut | din3sh: yes |
| 12:11.07 | kaldemar | change "allow=g729,ulaw,alaw,gsm" to separate allow lines. |
| 12:19.04 | bintut | kaldemar: ok, i will try that. |
| 12:28.45 | bintut | still, the speed of the sound is fast that i cannot understand and still getting the warning message when it dials to my analog channels: |
| 12:28.48 | bintut | [May 26 20:27:07] WARNING[26257]: channel.c:4913 ast_write: Codec mismatch on channel DAHDI/1-1 setting write format to g729 from ulaw native formats 0x4 (ulaw) |
| 12:28.49 | bintut | [May 26 20:27:07] WARNING[26257]: channel.c:4913 ast_write: Codec mismatch on channel DAHDI/4-1 setting write format to g729 from ulaw native formats 0x4 (ulaw) |
| 12:33.49 | bintut | and the transcoding of g729 codec does not work |
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| 12:37.38 | bintut | can anyone help me here? |
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| 13:00.08 | *** join/#asterisk joelsolanki (~joelsolan@123.237.172.246) |
| 13:00.16 | joelsolanki | Hello all. good morning. |
| 13:00.35 | joelsolanki | i have asterisk and i want to disable people dialing same number simultenously. |
| 13:01.26 | joelsolanki | for example if i dial 18158340675 it should process and same time other person dial 18158340675 i want to give me busy signal from my asterisk. |
| 13:02.51 | joelsolanki | so when i dial 1815840675 call processes but it is not connected with remote party. either in process or ringing but from my same switch other user dials same number i want to give him busy as that same number is in process or ringing from my side. |
| 13:02.54 | joelsolanki | any suggestions ? |
| 13:04.59 | WIMPy | Listen on AMI for dialplan events and keep a state of all channels. |
| 13:05.11 | WIMPy | But that will require some programming. |
| 13:05.25 | WIMPy | But depending on how you dial out it may even be impossible. |
| 13:05.56 | joelsolanki | ok |
| 13:06.28 | joelsolanki | i thought same solution would be there in dialplan :( |
| 13:24.20 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 13:24.30 | kami | Hello. |
| 13:27.40 | kami | WIMPy: ping |
| 13:29.32 | *** join/#asterisk dandate2 (~dan@180.191.227.172) |
| 13:29.50 | dandate2 | if i'm using g729 passthru and my music on hold is in .wav format do I need a license for each caller on hold? |
| 13:32.48 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 13:34.43 | WIMPy | kami: pong |
| 13:34.53 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:42.14 | kami | WIMPy: I saw your commits to lcr. Didn't realise that you're one of the devs some days ago. |
| 13:42.44 | kami | Thanks again for your help, but unfortunately, 'One-way Audio' is still a problem. |
| 13:43.15 | kami | I /think/ that the randomness which I described, is in fact not random. |
| 13:43.19 | WIMPy | Well, I'm not in to mISDN at all. I jjust did some patches for LCR. |
| 13:43.31 | WIMPy | That should help. |
| 13:43.59 | kami | In any case, you're an expert! :) |
| 13:44.28 | WIMPy | Oh. That was easy :-) |
| 13:44.38 | kami | When the SIP callee hangs up the line, then the next call has 'one-way audio' |
| 13:45.03 | kami | When the ISDN caller hangs up, the next call is OK. |
| 13:46.34 | WIMPy | But you got the issue without Asterisk as well, or is that no longer the case? |
| 13:48.48 | kami | WIMPy: that seemed to be an intermittent network problem. |
| 13:49.09 | WIMPy | Oh |
| 13:49.29 | WIMPy | Ok, so if it only happens with Asterisk, that smells like an import/export issue. |
| 13:50.51 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-219.bois.qwest.net) |
| 13:51.58 | kami | My symptoms are slightly different than these: http://thread.gmane.org/gmane.linux.isdn.i4l.user/5929/focus=5936 |
| 13:52.25 | kami | I have HFC USB, as I mentioned (an X-tension device) |
| 13:53.08 | kami | And the 'one-way' is the other way round: the ISDN caller cannot hear the SIP phone. |
| 13:54.48 | WIMPy | Have you tried to find out what is required to make it work again? |
| 13:56.16 | *** join/#asterisk xoveruk (~rich@host109-158-8-12.range109-158.btcentralplus.com) |
| 13:56.37 | xoveruk | hi i am getting 2569 traffic and not 4569 iax traffic on my trunks do you know what might be the cause of this? |
| 13:57.06 | WIMPy | The thing is that Audio does not go through LCR. chan_lcr will connect directly to mISDN for that. |
| 13:57.47 | xoveruk | and what is sonuscallsig? |
| 13:57.49 | WIMPy | xoveruk: A configured port number? NAT? |
| 13:59.36 | xoveruk | I am receiving it directly from the other asterisk |
| 13:59.54 | xoveruk | that is the source port and its sent my destination IP with port 4569 |
| 14:00.11 | xoveruk | usually I would see sr.4569 dst.4569 and the trunk works |
| 14:03.38 | kami | WIMPy: I tried to test systematically and see this behaviour: after restarting asterisk and lcr, the very first call which arrives for an MSN is mute. |
| 14:03.55 | kami | WIMPy: the second call however is OK. |
| 14:04.04 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:04.16 | kami | And if the ISDN caller hangs up, then the next call is also OK. |
| 14:04.21 | WIMPy | Ok, that seems to be another thing again. |
| 14:05.12 | kami | But if one of the calls is hanged up by the SIP callee, then the next call has one-way audio in the sense that the ISDN caller cannot hear the SIP callee. |
| 14:05.32 | xoveruk | can anyone assist in IAX2 troubleshooting? |
| 14:05.54 | WIMPy | I'm just updating to all current GIT versions and see if I get similar issues. |
| 14:06.06 | xoveruk | what is sonuscallsig? |
| 14:06.14 | kami | You probably don't have a USB HFC card, do you? |
| 14:06.18 | xoveruk | Im gettting a tonne of traffic of that type and no iax traffic |
| 14:07.12 | WIMPy | kami: I will try with the HFC-8S first, but I've got USB versions somewhere. |
| 14:08.18 | WIMPy | Does it make a difference who initiates the call? |
| 14:09.18 | kami | Oh, I'm not sure whether I tested that. Hang on. |
| 14:20.46 | xoveruk | jees, what is sonuscallsig? |
| 14:28.45 | WIMPy | kami: I get it the other way round. No Audio to SIP after the ISDN side hung up. |
| 14:29.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 14:29.32 | kami | xoveruk: WIMPy somehow already answered your question: 'A configured port number?' |
| 14:30.09 | WIMPy | Oops. That's actually a phone issue :-( |
| 14:30.17 | kami | xoveruk: port 2569 is registered with IANA for the 'Sonus Call Signal' |
| 14:30.26 | WIMPy | No, It's working. |
| 14:31.21 | WIMPy | So maybe it's the USB driver that hase the same or a similar issue. |
| 14:31.37 | kami | WIMPy: thank you for trying. |
| 14:32.20 | bintut | anybody can help me here? kindly check => http://www.pastie.org/3970890 |
| 14:32.32 | kami | I also tested again in the mean time and this time, I have the 'SIP voice has pauses of random length' issue again! |
| 14:32.45 | kami | It's driving me crazy. |
| 14:33.03 | kami | Let me test Echo() again. |
| 14:35.22 | WIMPy | loves the way you can just add an interface while the hole stuff is running. |
| 14:37.23 | kami | Echo works flawlessly. |
| 14:38.37 | kami | Which means that I'm back to square one. |
| 14:39.51 | kami | WIMPy: you said that chan_lcr talks to misdn directly. So, I would have to debug misdn/chan_lcr. Do you have any pointer to the debug options of the drivers? |
| 14:40.09 | WIMPy | Ok, I get the same issue with the USB dogle. |
| 14:40.23 | *** join/#asterisk bintut (~bintut@cm102.sigma15.maxonline.com.sg) |
| 14:40.30 | kami | WIMPy: ahhh! At least, I'm not crazy! |
| 14:41.28 | kami | The three modules are currently started with debug=0x0. What should I place there to get a reasonable trace of what happens? |
| 14:41.59 | WIMPy | I have no idea :-( |
| 14:42.20 | kami | Let me check the mailing list. |
| 14:42.28 | WIMPy | But it looks like I have to restart both Asterisk an LCR to get audio agqain. |
| 14:44.55 | kami | WIMPy: I sometimes have no audio for a long time. And while testing, one might hang up too early and assume that there is no audio at all, although you would hear something after a minute or so. |
| 14:45.35 | WIMPy | That sounds fishy. |
| 14:45.43 | *** join/#asterisk bintut (~bintut@cm102.sigma15.maxonline.com.sg) |
| 14:47.00 | bintut | anyone can help me on the g729 codec problem? |
| 14:48.01 | bintut | wonders if digium provide support for the g729 codec license bought from them |
| 14:50.00 | WIMPy | kami: When I call in via ISDN to LCRs MOH, I get short drop outs from time to time. |
| 14:50.24 | WIMPy | So it must be in the USB driver somewhere. |
| 14:53.58 | WIMPy | Yes, definitely a hfcsusb issue. |
| 14:56.56 | *** join/#asterisk bintut_ (~bintut@cm102.sigma15.maxonline.com.sg) |
| 14:58.20 | bintut | hello? |
| 14:58.53 | *** join/#asterisk bintut_ (~bintut@cm102.sigma15.maxonline.com.sg) |
| 14:59.13 | WIMPy | bintut_: I'm pretty sure they will. |
| 14:59.59 | WIMPy | does some shopping. BBIAB |
| 15:03.15 | kami | WIMPy: with debug=8, I get this in the syslog: ERROR stop_iso_gracefull for fifo 0.0 (and 0.1) |
| 15:03.35 | kami | WIMPy: have fun. |
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| 15:52.45 | *** join/#asterisk olinux (~olinux@208-104-179-59.fttp.sta.comporium.net) |
| 15:54.57 | olinux | remote polycom phones register and can receive/place calls but no sound once connected |
| 15:55.11 | olinux | it seems the server is seeing the remote lan ip |
| 15:55.17 | olinux | rather than the wan ip? |
| 15:56.57 | WIMPy | kami: From the hfcsusb? |
| 15:57.26 | WIMPy | olinux: That is perfectely normal. See |
| 15:57.33 | WIMPy | ~sipnat |
| 15:57.33 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 15:58.24 | olinux | thanks! reading now |
| 15:58.57 | kami | WIMPy: yes, it says 'HFC-S_USB.1: ERROR stop_iso_gracefull for fifo 0.0' |
| 15:59.11 | kami | and another line with 0.1 at the end |
| 15:59.39 | kami | This is a difference which I can see between the first and the subsequent calls. |
| 16:00.36 | kami | When I crank up the debug level, strange things happen (e.g. I had one reboot without anything in the syslog). |
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| 16:38.38 | bintut | anyone here familiar in using the g729 codec from digium on asterisk v1.8? |
| 16:41.54 | WIMPy | kami: I just reverted back to kernel 3.1.7 version. That's ok. |
| 16:42.09 | WIMPy | But needs a patch to recognize the X-Tensions dongle. |
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| 17:01.29 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 17:21.48 | WIMPy | Same for 3.4 |
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| 17:52.11 | bintut | what does this mean? => [May 27 01:48:02] WARNING[28005]: translate.c:256 ast_translator_build_path: No translator path from alaw to unknown |
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| 17:59.02 | mrkiko | Hi all! |
| 17:59.46 | mrkiko | Sorry for my rudeness, but I'm observing somevery preoccupying fenomena running asterisk as shipped by debian, Version: 1:1.8.11.1~dfsg-1 |
| 17:59.58 | mrkiko | but I checked Debian patches and init script without finding valid reasons |
| 18:00.01 | mrkiko | still |
| 18:00.11 | mrkiko | when launching the asterisk executable |
| 18:00.17 | mrkiko | or runningthescript |
| 18:00.20 | mrkiko | two main things happen |
| 18:00.55 | mrkiko | 1 - The file /dev/null suddenylbecome readable and writable by root only |
| 18:01.02 | mrkiko | 2 - Files saved in /dev/shm from now on, disappears |
| 18:01.14 | mrkiko | This system is a 32-bit system, with kernel 3.4.0 |
| 18:01.23 | mrkiko | are these behaviours known? |
| 18:01.53 | WIMPy | has never heard of anything like that. |
| 18:04.28 | mrkiko | also... and uses asterisk since some years now |
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| 18:09.27 | mrkiko | beaware guys !! :D:D |
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| 18:09.35 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 18:10.00 | atan | Okay so I moved Asterisk to a new box, how some of my devices won't connect correctly. But, 1 client is connected, without issue, and they are using a grandstream ata... they work fine, so I know things are in check |
| 18:10.34 | atan | I can't reason what might need to be changed to allow the others to work though. They're all loaded via realtime, and like I said one is working =\ |
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| 18:11.11 | atan | I do notice calls from the troubled phones shows chan_sip.c:23316 handle_request_invite: Call from '' rejected, which isn't right |
| 18:16.27 | ChannelZ | no identity |
| 18:20.12 | atan | ChannelZ, and how might I fix that? everythin is the same except for the host machine. configs are copies =\ |
| 18:21.45 | atan | At first I thought it may be a database issue somewhere with odbc, but since 1 client can connect and function fine I'm not sure. |
| 18:25.46 | atan | The other oddball is when the phone attempts to register it gets put into realtime, host=(Unspecified) when I use sip show peers. sip show peer shows the correct details from realtime but nothing unique to the phone like ip, or device |
| 18:26.16 | atan | But it's in there. Shows in the list as a cached realtime peer. |
| 18:28.45 | [TK]D-Fender | atan: Device isn't sending the INVITE properly. You have misconfigured it. |
| 18:32.48 | atan | [TK]D-Fender, it blows my mind that it was working prior. From what I can tell all that changed was the host machine for Asterisk. Does this provide anything useful? http://pastebin.com/7K60t5qU |
| 18:33.30 | atan | It is responding with SIP/2.0 401 Unauthorized from what I can see |
| 18:34.35 | [TK]D-Fender | if that is all then the device isn't responing to the auth challenge |
| 18:35.57 | atan | There has got to be something on the server I could change. That's all that I updated O_O |
| 18:40.46 | atan | Actually come to think of it. The last Asterisk install was on a VPS. This is on dedicated hardware, perhaps something fooy is going on with the NAT? |
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| 18:48.46 | bintut | [TK]D-Fender: can you help me regarding g729 codec? the information of my home pbx is at http://www.pastie.org/3970890 |
| 18:54.47 | bintut | in my "sip set debug on" output i got this => translate.c:256 ast_translator_build_path: No translator path |
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| 19:32.59 | bintut | waves |
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| 20:49.53 | twanny796 | installed and uninstalled and installed again, on debian. Now the configuration files did not get installed?!! |
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| 20:54.15 | atan | twanny796, did you use a package manager or compile from source? |
| 20:54.49 | twanny796 | atan: a package manage on debian |
| 20:55.38 | atan | use purge when removing, install again |
| 20:56.42 | twanny796 | atan: I did that |
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| 20:57.34 | atan | Then install from source and use make samples I believe it is :) |
| 20:58.09 | twanny796 | atan: maybe, but that's too much work! |
| 21:00.12 | atan | You came seeking a magic 2 word command to fix everything? |
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| 21:20.01 | shadebob | Hi. I have a "media gateway" with Debian+Asterisk 1.8.x + TE820B. No phones registered on it. I have a "registrar gateway" with Elastix and 400 phones connected to it. The 2 servers are connected throught an IAX trunk. How phone 1234 can use his outgoing DID (ex : 7076) on my PRI from my IAX link ? |
| 21:20.54 | WIMPy | The same way as always. Asterisk doesn't change the caller ID unless you explicitely tell it to. |
| 21:22.52 | [TK]D-Fender | And the terms "registrar gateway", "media gateway" are entirely inaccurate. |
| 21:23.06 | [TK]D-Fender | * is not a "split-duty SIP server., it is a B2BUA |
| 21:23.16 | [TK]D-Fender | So you are just passing calls from one to the other |
| 21:23.38 | shadebob | Thanks WIMPy and TK. |
| 21:23.44 | [TK]D-Fender | And to do that you set up boring peers just like absolutely everything else |
| 21:24.02 | shadebob | TK this is why I put quotes ;) |
| 21:25.39 | [TK]D-Fender | yes so you could highlight misapproprated words. Excellent way to draw attention to them. I might have missed them otherwise :p |
| 21:26.31 | WIMPy | Well, Asterisk and terminology is an interesting area anyway. And in #asterisk even more so. |
| 21:26.44 | shadebob | Last question : I have DL380 with 24Gb. Can I install 2 TE820B in my server or I will have problems with interrerupts for example …. |
| 21:27.02 | shadebob | 1 CPU quad core |
| 21:27.40 | WIMPy | 3 of those cores won't help with hardware interfaces. |
| 21:28.49 | WIMPy | My observation was that performance increases slightly when yu explicitely use only one core. |
| 21:29.26 | shadebob | even with 16 E1 ? |
| 21:29.37 | WIMPy | Doesn't matter. |
| 21:29.56 | WIMPy | But it might help if they use different drivers. |
| 21:31.59 | [TK]D-Fender | What does DIGIUM say about using multiple cards? |
| 21:32.48 | shadebob | No idea |
| 21:32.49 | shadebob | Currently I have an Elastix with 250 posts and 8 E1 ... Next week I have to "explode" here with 16 E1 and 400 phones. I am a little scared :) |
| 21:33.09 | WIMPy | You probably should put a "timing cable" on. But that will only work if all lines are from the same telco. |
| 21:33.51 | shadebob | 3 telco .... |
| 21:34.21 | WIMPy | Do not connect different telcos to the same card. |
| 21:36.00 | shadebob | Currently my three operators are connected to a unique TE820B ... No problems ... You think the timer can cause problem ? |
| 21:36.03 | WIMPy | TBH I'm not even sure what happens wit multiple lines from the same telco if they are SHDSL lines, as have become the norm. |
| 21:36.20 | WIMPy | Yes. The cards accept only one timing source. |
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| 21:37.01 | shadebob | I'm in France. Every E1 are SHDSL |
| 21:37.02 | WIMPy | So if those telcos aren't in sync, you are likely to run into some issues. |
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| 21:41.10 | WIMPy | pong |
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| 21:42.37 | Cadey | ping dag-erdy pong-derladerdy? |
| 21:45.55 | shadebob | WIMPy … It seem I have actually no problem with my different telco on the same car because I NEVER send calls from telco1 to telco2 (like transfert)…. http://lists.digium.com/pipermail/asterisk-users/2011-March/260428.html |
| 21:46.55 | WIMPy | It doesn;t matter if it's the same call. |
| 21:47.21 | WIMPy | The issue is that all ports are read at the clock of the one that's selected as timing source. |
| 21:47.31 | WIMPy | (and written, off course). |
| 21:49.11 | WIMPy | The glitches that could occur when connecting a call from one telco to another that is not is sync is insignificant. At least for audio. |
| 21:49.34 | shadebob | In any case thank you for your ideas. I will send you an email to give you the result;) |
| 21:50.18 | WIMPy | Be aware that even if multiple telcos on one card (seem to) work no, it may fail at any time. |
| 21:51.09 | WIMPy | s/no,/now,/ |
| 21:51.36 | shadebob | sure … I will try to balance correctly telco between card ... |
| 21:52.07 | shadebob | I think I will just have a problem with my 2 new Rhino Channel Bank |
| 21:52.17 | WIMPy | Does anyone know for sure if the Sangoma cards suffer from the same issue? |
| 21:53.04 | WIMPy | They don't seem to use the dahdi timing source configuration. |
| 21:53.38 | shadebob | "All Sangoma cards operate using 1 single clock source." |
| 21:53.54 | shadebob | from http://wiki.sangoma.com/Wanpipe-Debugging |
| 21:53.59 | WIMPy | :-( |
| 21:54.40 | shadebob | "If at least one of the lines connected to a port on the card has a different clock signal than the others, this can cause the dma buffers to be filled at different times. When the hardware interrupt is signalled, buffers are filled, while others are not full. This scenario causes wanpipe overruns." |
| 21:56.16 | shadebob | It seem Patton gateway have timing clock per port |
| 21:56.24 | WIMPy | Not really feasable for real-world situations. |
| 21:57.06 | WIMPy | So there actually is one advantage with gateways. |
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| 22:04.33 | shadebob | WIMPy … Maybe fonebridge is the solution ... |
| 22:05.35 | shadebob | http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=38 |
| 22:06.33 | WIMPy | Not sure. What I read in here abot them makes me sceptic. |
| 22:06.50 | shadebob | not clear |
| 22:07.37 | WIMPy | People have been having some issues with them. |
| 22:07.48 | WIMPy | Off course I don;t know how many don't. |
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| 22:17.48 | jpsharp | I had the same clocking problem with an older Seimens switch. It couldn't resolve the differences between T1 clocking off two carrier. We ended up running all the circuits through some Adtran box to clock them together better. |
| 22:18.04 | jpsharp | But I'll be darned if I can't find what that box was called. |
| 22:18.24 | shadebob | interesting ! |
| 22:19.22 | WIMPy | I've been told that the Siemens thingies are very sensitive to timing glitches and sometimes have issues with only one carrier if it's delivered via SHDSL. |
| 22:21.06 | jpsharp | It was the same problem that is being described. The switch developed a master clock based off one circuit and to hell with clocking on the rest. Everything was fine until a circuit from a different carrier was brought in. |
| 22:22.01 | jpsharp | We could clock off any of the 3 circuits from AT&T and take frameslips (and thus noise, clicks, and disconnects) on the MCI circuit. Or clock off the MCI circuit and have lots of calls dropped on the AT&T circuits. |
| 22:22.12 | WIMPy | As already said: Not really feasable for real-world situations. |
| 22:22.42 | WIMPy | If I need more than one line, I'd certainly not take all from the same telco, |
| 22:24.25 | [TK]D-Fender | The only time we hear about Fonebridge is when they have problems. Which says something... |
| 22:24.50 | WIMPy | Does it? |
| 22:25.07 | WIMPy | Why should they tell us about them if they don't have issues. |
| 22:25.39 | [TK]D-Fender | Nothing good... |
| 22:25.45 | [TK]D-Fender | basically lots of trouble |
| 22:25.48 | shadebob | I have a customer with at least 10 fonebridge. I will ask him monday morning ... |
| 22:26.30 | WIMPy | We once had someone who just wanted to get rid of them and offered them for free. |
| 22:27.53 | shadebob | the only comment he made to me was on the consumption of bandwidth by TDMOE |
| 22:29.44 | shadebob | with more than 250,000 euros per month phone bill I think it should work: s |
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