| 00:05.05 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 00:05.16 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:09.47 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.170.249) |
| 00:20.57 | phix | Do I need to do anything speical to get a SIP extensions working as a member of a queue? |
| 00:21.10 | phix | Do I need to register to the queue first? or can I automagically do this? |
| 00:25.57 | *** join/#asterisk cstachris (~chrismylo@42.241.106.79) |
| 00:31.17 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:33.43 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 00:37.00 | ChrisInSydney | phix: Will it always be in the queue ? |
| 00:41.17 | [TK]D-Fender | phix: Nothng special, and you register if your SIP device is supposed to register. |
| 00:41.29 | [TK]D-Fender | It has nothing to do with being a member of a queue or not |
| 00:49.29 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 00:49.45 | *** join/#asterisk rampage73 (~rampage73@bob.dctechonline.com) |
| 00:50.36 | rampage73 | anyone know how I can disable call waiting per extension in Asterisk 1.6.2.5-0ubuntu1.4 ? |
| 00:55.29 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:56.10 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:56.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:56.54 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 00:59.04 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 00:59.47 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 01:11.23 | *** join/#asterisk jasonwert (~w3rt@24-236-215-82.dhcp.trcy.mi.charter.com) |
| 01:11.39 | [TK]D-Fender | rampage73: What device are you using that you see the existance of "call waiting" for? |
| 01:14.26 | rampage73 | hey [TK]D-Fender , if i understand the question correctly the device is a spa2102 , I know I can disable callwaiting if i could log into it (that requires a drive) I thought there was a way to just disable it in asterisk (and save the drive at 3.85 a gallon ;) |
| 01:14.51 | rampage73 | [TK]D-Fender, if i misunderstood please enlighten me |
| 01:15.49 | [TK]D-Fender | rampage73: * knows nothing of "call waiting" for that device/ |
| 01:16.04 | [TK]D-Fender | rampage73: It is just a SIP device and it either accepts another call or it doesn't |
| 01:16.23 | rampage73 | [TK]D-Fender, ok so driving it is! thanks |
| 01:16.27 | [TK]D-Fender | Before actually dialing it you COULD check the DEVICE_STATE() for it and choose not to dial it... |
| 01:17.16 | rampage73 | [TK]D-Fender, hmm that is a good idea I may check into that Thanks again. |
| 01:19.26 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 01:23.04 | *** join/#asterisk Twitchnln (~Adium@c-98-242-79-16.hsd1.ga.comcast.net) |
| 01:23.19 | *** part/#asterisk Twitchnln (~Adium@c-98-242-79-16.hsd1.ga.comcast.net) |
| 01:25.36 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:45.28 | *** join/#asterisk nargon (1000@adsl-76-226-250-177.dsl.sfldmi.sbcglobal.net) |
| 01:46.29 | nargon | hi, I have broadvoice sip peer on context mikeA, doing a show sip peer on the sip line shows the proper context of mikeA |
| 01:47.04 | nargon | however when I dial that public broadvoice number, asterisk is saying 'rejected extension not found in context mikeB |
| 01:47.31 | nargon | this just started happening, not sure why calls to a sip line with one context is showing up as being processed in another context... |
| 01:50.19 | [TK]D-Fender | nargon: Show us your configs and your call and we'll tell you why.... |
| 01:50.22 | [TK]D-Fender | ~pb |
| 01:50.23 | 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. |
| 01:50.24 | [TK]D-Fender | ^^^ |
| 01:54.03 | *** join/#asterisk pdtpatrick (~ptaylor@12.249.4.226) |
| 01:54.14 | nargon | http://pastecode.com/s4 |
| 01:58.49 | nargon | you can see the sip peer [bvUSWICOM] has its context set to "from-USWiCom-Mike" |
| 01:59.35 | nargon | but odly the output on console shows its being processed under "from-MIT-Mike" |
| 02:00.33 | nargon | and that same console output line shows the originating phone # as the line for "[bvMITAdminMike]" instead of my att landline number I used to call in with |
| 02:01.40 | nargon | sip show peer bvUSWICOM verifies the context is set appropriately for the trunk, as it is in the sip.conf |
| 02:05.24 | *** join/#asterisk lkthomas (~lkthomas@59.152.236.158) |
| 02:05.26 | lkthomas | hey guys |
| 02:05.40 | lkthomas | how could I show if the PRI connection have problem ? |
| 02:12.48 | [TK]D-Fender | nargon: Now show a call with actual SIP DEBUG enabled |
| 02:12.54 | [TK]D-Fender | nargon: sip set debug on |
| 02:13.23 | [TK]D-Fender | lkthomas: dahdi show span 1 |
| 02:13.27 | [TK]D-Fender | dahdi show status |
| 02:13.31 | [TK]D-Fender | etc |
| 02:20.50 | nargon | got 2 weed out all the ip addresses |
| 02:21.31 | [TK]D-Fender | Don't |
| 02:21.58 | nargon | ok |
| 02:23.56 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 02:27.02 | nargon | http://pastecode.com/s5 |
| 02:27.27 | lkthomas | provisioned down |
| 02:27.30 | lkthomas | seems telco problem |
| 02:28.21 | *** join/#asterisk RickCogley_ (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 02:31.26 | [TK]D-Fender | nargon: Found peer 'bvMITAdminMike' for '7347691442' from 206.15.148.221:5060 |
| 02:31.43 | [TK]D-Fender | nargon: Looking for 3017039705 in from-MIT-Mike (domain 173.225.186.34:5060) |
| 02:32.31 | [TK]D-Fender | nargon: You have 2 peers with the same HOST. Neither auth incoming calls. |
| 02:32.39 | [TK]D-Fender | nargon: it's first-come first served |
| 02:33.19 | nargon | http://pastecode.com/s5 |
| 02:33.23 | nargon | opps |
| 02:33.29 | [TK]D-Fender | nargon: So you're not going to separate them that way. You'r just going to have to use a common inbound context |
| 02:33.53 | nargon | hmm it used to work |
| 02:34.04 | [TK]D-Fender | It should always act as we see |
| 02:34.05 | nargon | but if the context is specified for the peer it should simply use that context |
| 02:34.17 | [TK]D-Fender | nargon: It's hitting the first pewer no matter what <- |
| 02:34.29 | [TK]D-Fender | nargon: It will never hit the other one |
| 02:34.37 | nargon | what is a pewer? |
| 02:34.39 | [TK]D-Fender | peer* |
| 02:34.42 | nargon | oh |
| 02:35.13 | nargon | but that defies the whole point of contexts |
| 02:35.23 | nargon | when a peer is tied to a context, it has to use that context in the dialplan |
| 02:35.33 | [TK]D-Fender | nargon: nargon Ony for the fact you have multiple peers to the same host |
| 02:35.49 | [TK]D-Fender | And the FIRST peer is a legitimate hit <- |
| 02:36.09 | [TK]D-Fender | it is acting normal. |
| 02:38.26 | nargon | so your saying because they both use broadvoice the context is not working properly |
| 02:39.23 | [TK]D-Fender | 1st peer matches. Doesn't have any reason to look further. There is nothing in the call identifying it in a way to differentiate |
| 02:39.30 | [TK]D-Fender | Context IS working. |
| 02:39.38 | nargon | ok what does '1st peer matches' mean |
| 02:40.08 | [TK]D-Fender | the first peer is MATCHED against the incoming call. It MATCHES'\ |
| 02:40.20 | [TK]D-Fender | nargon: right host? Yes? GO. |
| 02:40.33 | nargon | not sure what 'first peer' means yet |
| 02:40.56 | [TK]D-Fender | [bvMITAdminMike] <------------------- |
| 02:41.11 | nargon | what does it match? |
| 02:41.19 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:41.21 | [TK]D-Fender | host=sip.broadvoice.com <------------------------ |
| 02:41.29 | nargon | ahh |
| 02:41.37 | nargon | ok so thats causing it to circumvent context |
| 02:41.45 | [TK]D-Fender | I told you "same host" repeatedly |
| 02:41.51 | nargon | i heard you repeatedly |
| 02:41.54 | [TK]D-Fender | Doesn't circumvent |
| 02:42.15 | [TK]D-Fender | first = valid. matches as it should, goes to the context it should accordingly |
| 02:44.06 | nargon | so your saying instead of yeilding to the context specified in sip.conf, its first 'matching' based on the source hostname, which happens to line up with both peers |
| 02:44.25 | [TK]D-Fender | nargonso your saying instead of yeilding to the context specified in sip.conf, <- what is this you are going on about? |
| 02:44.28 | [TK]D-Fender | Yeilding? |
| 02:44.43 | [TK]D-Fender | There is nothing in the incoming call to cause it to have any reason to choose one over the other. |
| 02:44.44 | nargon | ok never mind i can tell you are upset |
| 02:44.52 | nargon | leave it |
| 02:45.02 | *** part/#asterisk nargon (1000@adsl-76-226-250-177.dsl.sfldmi.sbcglobal.net) |
| 02:45.14 | [TK]D-Fender | You seem insitent that it should amtch the other. There is simply no element that uniquely identifies it so that * could tell |
| 02:46.31 | [TK]D-Fender | yup... when in doubt just insist that * HAS to know...... |
| 02:46.34 | [TK]D-Fender | *sigh* |
| 02:55.06 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 03:26.28 | *** join/#asterisk RickCogley (~anonymous@EM1-112-41-70.pool.e-mobile.ne.jp) |
| 03:54.15 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:58.17 | *** join/#asterisk RickCogley (~anonymous@EM1-112-41-70.pool.e-mobile.ne.jp) |
| 03:59.05 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:34.19 | *** join/#asterisk cstachris (~chrismylo@114.73.65.20) |
| 05:10.55 | *** part/#asterisk Tyrael1 (~Ryan@c-50-129-214-142.hsd1.in.comcast.net) |
| 05:19.11 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 05:44.59 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:45.31 | joako | Is there a way to actually change the email address asterisk uses to send messages? |
| 05:47.40 | [TK]D-Fender | voicemail.conf <--- |
| 05:48.04 | joako | [TK]D-Fender, Yes and changing that has no effect. My mail is rejected because it is from asterisk@localhost |
| 05:49.44 | [TK]D-Fender | I don't see what you "changed" or how you've attempted to apply your changes, or the failure that follows. Pick an order and start rectifying that ;) |
| 05:52.15 | joako | If I change serveremail = in voicemail.conf the email is still sent by root@localhost |
| 05:52.54 | shihan | quite possibly its the unix side config not letting asterisk change its from address |
| 05:55.19 | phix | hey I am having issues making calls via my VOIP provider |
| 05:55.53 | phix | If I call the VoIP number I get through, the numbers ring but I can hear the other end, my first thought was firewall and NAT but I have the settings for that correct I am sure |
| 05:56.16 | phix | any thing else that would stop it from working correctly? like fromdomain sip config entry not matching? |
| 05:57.37 | phix | do I even need to set fromdomain? |
| 05:58.21 | [TK]D-Fender | That changes if they even accept your call... doesn't change audio] |
| 05:58.38 | [TK]D-Fender | And assuming you did everything right is the fastest way to make a mistake |
| 05:58.59 | [TK]D-Fender | That slips past you continuously |
| 06:01.46 | phix | hmmmm |
| 06:03.45 | phix | well I am using default RTP of 10000 - 20000, this is forwarded from my router to the correct internal IP, I have externhost and outside_addr set to correct hostname of the IP address that packets are originating from |
| 06:03.52 | phix | what else should I check? |
| 06:04.08 | [TK]D-Fender | EVERYTHING |
| 06:04.37 | phix | Not everything is involved in making a call though |
| 06:04.43 | joako | phix, I find that generally if everything else is correct no ports need to be forwarded |
| 06:04.50 | [TK]D-Fender | And you gave me statments, not actual proof of what you really did, where, and that it actually took. And that is all still hypotheical to what's actually happening in the actual call |
| 06:05.19 | phix | I didnt see any error messages when enabling sip debug |
| 06:05.26 | phix | I am guessing it is a RTP issue |
| 06:05.36 | phix | I am using a/ulaw |
| 06:06.04 | phix | I have enough bandwitdh, syncing at about 8-10Mbit |
| 06:08.28 | phix | now what [TK]D-Fender ? how can Ifind out what is going on? |
| 06:08.33 | phix | where can I look? |
| 06:09.04 | [TK]D-Fender | At the actual call. |
| 06:10.23 | phix | I try and send a DMTF and nothing goes through either, I can hear the queue music on hold music |
| 06:10.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:10.45 | phix | but if someone picks up no of us can hear the other |
| 06:10.51 | phix | no = none |
| 06:11.26 | phix | Would my VOIP provider do something stupid like force RTP ports and not tell me about it? or is there something more obvious I can check |
| 06:12.21 | [TK]D-Fender | Ok, this isn't going to be getting anywhere it seems .... |
| 06:13.21 | phix | ok so what you suggest [TK]D-Fender ? |
| 06:14.29 | [TK]D-Fender | I've said it 3 times already. I'm not interested in spinning this in circles. |
| 06:15.42 | phix | No you didnt answer me at all |
| 06:16.40 | phix | repaste as I cant see any answer above |
| 06:17.18 | phix | thnx |
| 06:20.16 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 06:37.13 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-veezbuxtgadfwhpk) |
| 06:43.04 | [TK]D-Fender | Well that's it for tonight |
| 06:43.26 | phix | aawww |
| 06:43.37 | phix | You need a nap |
| 06:44.31 | [TK]D-Fender | Over hear they call it "sleep". So far only 5.5 hrs of it coming |
| 07:05.05 | phix | aawww |
| 07:05.05 | phix | [Mar 1 17:54:45] WARNING[6567]: ast_expr2.fl:445 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input: |
| 07:05.09 | phix | ""GivenName Surname"<sip:5551234567@my.sip.provider.tld>" =~ "5551234567" |
| 07:09.41 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:15.29 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 07:19.31 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:20.13 | v0lZy | morning. |
| 07:24.16 | *** join/#asterisk irieKEN (~ken@cpe-72-130-175-87.san.res.rr.com) |
| 07:34.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:34.32 | schmidts | good morning |
| 07:34.37 | wdoekes2 | morning |
| 07:36.28 | irieKEN | morning. |
| 07:36.36 | kaldemar | phix: it's the "" around GivenName Surname that causes the parser to confuse. |
| 07:37.12 | kaldemar | phix: which function/variable are you feeding to the expression? |
| 07:42.22 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 07:45.00 | *** join/#asterisk AviMarcus (~avi@bzq-79-177-136-150.red.bezeqint.net) |
| 07:45.07 | AviMarcus | Anyone having trouble calling TellMe? |
| 07:53.43 | *** join/#asterisk srbh (~srbh@124.123.173.211) |
| 07:55.30 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:59.34 | ChrisInSydney | Hi all. Any suggestions for hardware specs for an Asterisk system for 30 simultaneous g711 calls with call recording on all channels ? |
| 08:00.05 | ChrisInSydney | AviMarcus: I thought M$ shut down the SIP URI for Tellme |
| 08:00.40 | AviMarcus | I have 3 different 1-800 carriers in my routes and all get rejected |
| 08:01.02 | AviMarcus | ChrisInSydney, you'll be more limited by disk IO than cpu I think. |
| 08:02.37 | ChrisInSydney | just connected to TellME via our local carrier here, They use ibasis for wholesale international |
| 08:02.48 | ChrisInSydney | 18005558355 |
| 08:03.22 | AviMarcus | 2012-03-01 10:02:52.742597 [NOTICE] sofia.c:6293 Hangup sofia/external/18005558355@sip.tollfreegateway.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] |
| 08:03.47 | AviMarcus | 2012-03-01 10:02:52.992573 [NOTICE] sofia.c:6293 Hangup sofia/external/99139150018005558355 [CS_CONSUME_MEDIA] [CALL_REJECTED] |
| 08:03.48 | AviMarcus | weird. |
| 08:04.09 | ChrisInSydney | Re recording, correct. 30 calls I figured I would need SSD, but now sure how many to stripe |
| 08:04.34 | AviMarcus | planning on compressing to mp3 or something? major cutdown on disk space required. |
| 08:05.44 | AviMarcus | even so.. one normal HD should be able to handle it. |
| 08:05.46 | ChrisInSydney | would be transferring and compressing off the system |
| 08:05.53 | ChrisInSydney | Cool |
| 08:06.05 | AviMarcus | well, see 30 * your encode speed |
| 08:06.08 | AviMarcus | erm rate |
| 08:06.24 | AviMarcus | 5meg a second? I get 20-30mb write speed on old HDs over USB |
| 08:06.31 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:06.43 | ChrisInSydney | SATA2 SSD should cope |
| 08:06.47 | AviMarcus | lol |
| 08:06.59 | AviMarcus | one of those 500mb write/second? That's what's in this machine ^^ |
| 08:07.50 | ChrisInSydney | cool, havent done much above 10 at a time,so just wanted to make sure |
| 08:08.05 | bulkorok | record to memdisk and write to hdd from there... |
| 08:08.09 | AviMarcus | I haven't done much above 2 :x the numbers seem pretty small though. |
| 08:08.28 | bulkorok | but 30 channels are not that much ^^ |
| 08:08.30 | ChrisInSydney | bulkrock, that was my other thought |
| 08:08.56 | ChrisInSydney | maybe set up a couple of Gb RAM disk |
| 08:09.21 | bulkorok | yeap... ram is not that expensive these days... |
| 08:09.49 | ChrisInSydney | true |
| 08:10.03 | ChrisInSydney | Thanks guys, must go and get dinner |
| 08:10.10 | ChrisInSydney | catch dinner |
| 08:10.20 | AviMarcus | hmm avg 240meg/second read rate on my ssd. that's disappointing. |
| 08:10.24 | ChrisInSydney | its fish n chips tonight |
| 08:11.05 | bulkorok | ew... it's 09:00 in the morning here and you talk about fish'n'chips Oo |
| 08:11.18 | ChrisInSydney | yup |
| 08:11.52 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 08:12.37 | ChrisInSydney | back later. Thanks again |
| 08:16.10 | AviMarcus | geez it seems it's systemic, I tried 4 various 1-800 TF numbers and none connected on all 3 carriers. WTF. |
| 08:23.35 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:32.17 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:34.51 | *** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it) |
| 08:38.18 | *** join/#asterisk cstachris (~chrismylo@114.73.102.207) |
| 08:38.35 | AviMarcus | switched 1st route to tf.callwithus.com seems to work now. weird. |
| 08:38.46 | k3asd` | hi |
| 08:41.06 | *** part/#asterisk AviMarcus (~avi@bzq-79-177-136-150.red.bezeqint.net) |
| 09:09.33 | *** join/#asterisk k3asd` (~k3asd@host168-86-dynamic.21-87-r.retail.telecomitalia.it) |
| 09:10.29 | *** join/#asterisk briankb (~briankb@pool-98-119-173-84.lsanca.fios.verizon.net) |
| 09:28.32 | *** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru) |
| 09:35.02 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:38.19 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:38.52 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 09:43.25 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:48.59 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:58.03 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 10:15.23 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:21.50 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:22.48 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 10:29.18 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 10:30.26 | Dovid | hi all. i have two asterisk servers sending calls between each other with OpenSipS in the middle (which should not affect the RTP). The issue I an having is that for some reason the asterisk machines change the coded they are using constantly. from G729 to G711A back to G729 etc. |
| 10:34.32 | Dovid | http://pastebin.com/12j14eDu |
| 10:34.40 | Dovid | any reason why asterisk would keep flip flopping? |
| 10:40.14 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 10:42.16 | kaldemar | Dovid: you might want to attach some sip debug. |
| 10:44.58 | Dovid | kaldemar: I don't see anything going on in asterisk as it flips the codecs but i will enable. one sec |
| 10:45.59 | Dovid | kaldemar: I did sip debug and nothing added in when it was flipping codecs |
| 10:47.15 | kaldemar | does it behave the same if you drop the OpenSIPS from between? |
| 10:49.43 | Dovid | http://pastebin.com/mJsjTQ1W |
| 10:50.46 | Dovid | kaldemar: Can't drop it. must go through OpenSipS. either way OpenSipS does not touch the media. it seems that asterisk is changing the codec and says: [2012-03-01 11:45:27] DEBUG[24645] res_rtp_asterisk.c: Created smoother: format: g729 ms: 20 len: 20 i wonder why that is |
| 10:51.22 | Dovid | also i saw some bug on rtcp. i wonder if i can siable that for now |
| 10:52.02 | *** join/#asterisk wonderworld (~ww@dsdf-4db5143f.pool.mediaWays.net) |
| 10:53.14 | *** join/#asterisk DarkRift (~dark@jenkins/developer/DarkRift) |
| 10:54.42 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 10:55.26 | kaldemar | Dovid: what are you trying to do with SIP_CODEC=g729 there? |
| 10:55.33 | hurdman | hi |
| 10:55.44 | hurdman | what does it mean : wct4xxp 0000:84:08.0: TE410P: RECEIVE slip NEGATIVE on span 2 |
| 10:55.58 | hurdman | i have never seen that before on my dahdi debug |
| 10:57.12 | Dovid | kaldemar: to force g729 on the call so it should not flip flop |
| 10:58.02 | kaldemar | Dovid: SIP_CODEC only sets codec for an inbound call. SIP_CODEC_OUTBOUND is what you'd want to set. if you don't try to set the variable, does it still keep changing it? |
| 10:59.52 | kaldemar | hurdman: you got a timing slip. |
| 11:02.07 | hurdman | kaldemar: how can i suppress that ? |
| 11:04.30 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:04.38 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 11:04.40 | kaldemar | hurdman: are you getting is constantly or was it just a single occurrence? |
| 11:05.19 | hurdman | regulary |
| 11:07.18 | hurdman | my timing source have a problem ? |
| 11:07.23 | kaldemar | looks like it. |
| 11:07.28 | hurdman | ergh |
| 11:08.23 | kaldemar | what are your timing settings in system.conf and what are your spans connected to? |
| 11:08.54 | *** join/#asterisk k3asd` (~k3asd@host168-86-dynamic.21-87-r.retail.telecomitalia.it) |
| 11:09.30 | hurdman | kaldemar: i have two TE420P |
| 11:11.14 | hurdman | http://pastebin.com/EZFJQH16 <= something like that in my system.conf |
| 11:12.07 | Dovid | kaldemar: I only set it because of the problem. i did an rtp debug. it shows the asterisk keeps flipping but still gives no reason as to why it keeps chaging the codec |
| 11:12.11 | kaldemar | hurdman: something like that or those settings exactly? |
| 11:14.53 | hurdman | thos settings :) |
| 11:15.01 | hurdman | i juste drop the last "group" |
| 11:15.16 | hurdman | ( 8 ) |
| 11:16.54 | hurdman | kaldemar: it looks false ? |
| 11:17.22 | v0lZy | hello everyone |
| 11:17.28 | v0lZy | think i got it working with static ports. |
| 11:17.31 | v0lZy | 1 more question. |
| 11:17.32 | hurdman | hi v0lZy |
| 11:17.59 | v0lZy | im using a sip trunk with many numbers... but outgoing number that displays on the person's screen im calling is always the same |
| 11:18.04 | kaldemar | hurdman: what is span 2 connected to? |
| 11:18.12 | v0lZy | how can i set it so that it changes based on the internal number thats calling out? |
| 11:18.48 | hurdman | a T2 |
| 11:19.27 | kaldemar | v0lZy: make sure your ITSP allows you to do it and then just set caller id to a valid number on outgoing calls. |
| 11:19.57 | ChrisInSydney | v0lZy: Hi, |
| 11:19.59 | v0lZy | kaldemar: i had to configure the number in "username" field |
| 11:20.02 | v0lZy | hi ChrisInSydney |
| 11:20.16 | v0lZy | but the thing is i have more than 1... and now whenever ic allout, it uses the one i entered in the username |
| 11:20.24 | v0lZy | i have an option to set outgoing caller id for each phone account... |
| 11:20.28 | ChrisInSydney | v0lZy: Did you jump on the VUC call in ? |
| 11:20.29 | v0lZy | but thatdoesnt seem to solve anything |
| 11:20.49 | v0lZy | ChrisInSydney: No, i didnt get the chance to. girlfriend dropped by |
| 11:20.50 | kaldemar | hurdman: i'd set timing priority to 1 on the span that is connected to your provider. |
| 11:20.52 | hurdman | kaldemar: i compare my conf with an older server, it's different, i make some test and back on irc |
| 11:21.17 | hurdman | kaldemar: yes it looks like the conf haven't been modified since new T2 was plug on |
| 11:21.24 | ChrisInSydney | v0lZy: Much better option ! |
| 11:21.41 | v0lZy | girlfriend? we had an argument |
| 11:21.46 | kaldemar | v0lZy: you're not listening. ask your provider if you have a range of numbers you can use as caller id number when dialing out. then set the caller id in your dialplan, leave your GUI alone. |
| 11:21.54 | v0lZy | last 30 days have been hell for me in almost all areas. |
| 11:22.24 | v0lZy | just received a huge blow by our half assed 'justice' (khm) system... |
| 11:22.37 | v0lZy | lost a case at court where i was sued ... |
| 11:22.39 | ChrisInSydney | v0lZy: We are only good for Asterisk advice here, not realtionship advice, did you try to restart ? Can you pastebin the argument ? |
| 11:22.46 | ChrisInSydney | :D |
| 11:23.12 | v0lZy | kaldemar: ok, let me cal lthem |
| 11:23.16 | ChrisInSydney | Ouch: THere is no justice, only law |
| 11:23.44 | v0lZy | well, funny thing |
| 11:23.54 | v0lZy | the court took it upon itself to be 'just' |
| 11:24.07 | v0lZy | and the whole thing had nothing to do with law for a change. |
| 11:24.16 | v0lZy | to give an example |
| 11:24.21 | ChrisInSydney | Yu mean they just listened to their aguments and not yours |
| 11:25.13 | v0lZy | 'we have no reason to believe the witness proposed by the accusing side is lying when she is saying she gave her daughter 50 000 €' .... the witness is a social welfare case that lives in a social welfare rent apartment on a minimum pension of something like 350 eur a month. |
| 11:25.24 | v0lZy | and uh... there's no evidence that she did give the money. |
| 11:25.32 | v0lZy | but yeah... 'we have no reason to doubt her' |
| 11:25.52 | v0lZy | 'even if she's the prosecuting clients mother'. |
| 11:26.41 | ChrisInSydney | File an appeal and threaten to go to the press. Settlement time |
| 11:26.55 | v0lZy | i have 15 days to file an appeal. |
| 11:26.58 | v0lZy | which im doing of course. |
| 11:27.06 | ChrisInSydney | sounds complicated |
| 11:27.16 | v0lZy | but in this backwards country, id be surprised if it would even arrive on time if i brought it to them personally.. like a month ago. |
| 11:29.07 | ChrisInSydney | bummer |
| 11:29.48 | *** join/#asterisk gucki (~gucki@dslb-092-074-234-082.pools.arcor-ip.net) |
| 11:29.55 | gucki | hi there |
| 11:30.19 | ChrisInSydney | g'day |
| 11:30.19 | gucki | when using the asterisk 1.8 debian packages, shall i be able to use the ilbc codec? :) |
| 11:30.51 | ChrisInSydney | NFI :-/ |
| 11:31.03 | ChrisInSydney | sorry, build your own isnt that hard |
| 11:31.24 | ChrisInSydney | but I have never botehred with precompiled versioins |
| 11:31.43 | ChrisInSydney | versions |
| 11:31.58 | *** join/#asterisk RickCogley_ (~anonymous@114.179.8.185) |
| 11:32.14 | gucki | ChrisInSydney: yeah, but if I don't need to compile, why do it ;) |
| 11:32.23 | gucki | ChrisInSydney: mh, core show codecs audio lists ilbc codec |
| 11:32.36 | gucki | so I guess it should be usable |
| 11:32.39 | ChrisInSydney | then you are OK. |
| 11:32.41 | hurdman | kaldemar: re |
| 11:32.43 | ChrisInSydney | One would expect |
| 11:33.45 | hurdman | kaldemar: http://pastebin.com/2yTc1bjp <= i have made this, but same result |
| 11:33.47 | ChrisInSydney | gucki: You would compile so you get all the bits you want, and maybe the source is a little more recent |
| 11:34.47 | ChrisInSydney | gucki: I also have "tweaked" some of the c code to do things I need it to. Couldnt do that with a bin disty |
| 11:36.03 | ChrisInSydney | That said, I still grab most apps via yum on the CentOS boxes |
| 11:37.12 | hurdman | kaldemar: no i receive only wct4xxp 0000:84:08.0: TE410P: RECEIVE slip POSITIVE on span 4 none slip NEGATIVE |
| 11:41.09 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 11:42.02 | Dovid | i guess i will wait for the devs to come online |
| 11:43.11 | gucki | ChrisInSydney: mh ok, it seems i now have to compile asterisk. ilbc has been removed from the source and must be added by the contrib scripts.. :( |
| 11:44.26 | ChrisInSydney | gucki: untar to /usr/src; ./configure; make menuconfig; make; make install |
| 11:44.40 | ChrisInSydney | make samples if you need |
| 11:45.05 | ChrisInSydney | hint: make -j4 will use 4 processor threads, nice on a multi core / cpu system |
| 11:45.47 | ChrisInSydney | Dovid: Have you tried just a single codec in your sip.conf stuff |
| 11:45.49 | ChrisInSydney | ?? |
| 11:45.52 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 11:46.16 | hurdman | kaldemar: i have tried some over conf, i think i don't understand or i miss something |
| 11:46.56 | hurdman | kaldemar: my last test is : http://pastebin.com/Z14h42qe |
| 11:49.47 | kaldemar | hurdman: 0 means that the span provides timing, a non-zero value means that the span takes timing from the line. you should never try to provide timing on a span that is connected to a provider. |
| 11:50.30 | hurdman | now i have span 1 , 2 , 3 and 4 connecting to a provider |
| 11:51.01 | hurdman | the others are connected to test machine |
| 11:51.45 | gucki | ChrisInSydney: yeah thanks, compiling right now :) |
| 11:51.54 | ChrisInSydney | cool |
| 11:52.59 | ChrisInSydney | Dovid: Are you using SIPS as a border controller or to jump NAT ? |
| 11:53.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:55.06 | hurdman | kaldemar: so something like that sounds better ? http://pastebin.com/cnQM9MsF |
| 11:56.00 | kaldemar | hurdman: that looks better. |
| 11:56.04 | hurdman | ok |
| 11:56.40 | hurdman | i ever get slip POSITIVE on span on 3 and 4 , and slip NEGATIVE on span 2 |
| 11:56.53 | hurdman | it's not the same provider on spam 1/2 and spam 3/4 |
| 11:57.00 | hurdman | i should change something ? |
| 11:57.28 | hurdman | ( never on 1 ) |
| 11:58.46 | kaldemar | ouch. you have a worst case scenario there if the providers' clocks are not in sync. |
| 11:59.13 | hurdman | arf |
| 12:00.22 | hurdman | can i set all to 1 ? |
| 12:01.57 | WIMPy | Ah, cool. I always wondered if that is possible on the Digium cards. |
| 12:03.46 | kaldemar | as far as i know DAHDI only selects one as master clock, so setting them all to 1 will just cause some confusion or makes it select the master clock in some way. |
| 12:04.16 | hurdman | kaldemar: so there's nothing i can do better than my last conf ? |
| 12:04.25 | WIMPy | That's my fear as well. |
| 12:04.32 | kaldemar | hurdman: as far as i know, you can't. |
| 12:04.35 | hurdman | <PROTECTED> |
| 12:04.47 | hurdman | kaldemar: thanks for your help ;) |
| 12:05.16 | WIMPy | Use smaller cards and no timing cable. |
| 12:05.37 | WIMPy | wonders if the Sangoma cards are able to manage that situation. |
| 12:05.49 | hurdman | it's the first time i get this error |
| 12:06.07 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:06.12 | hurdman | and i have ever 2 server with 4 card and 3 different provider /o\ |
| 12:06.17 | kaldemar | WIMPy: i guess DAHDI might be a limiting factor there also. |
| 12:06.21 | hurdman | ( 4 card per server ) |
| 12:06.49 | WIMPy | I had the impression that the dahdi timing config isn't used with Sangoma. |
| 12:07.20 | WIMPy | The only safe way is not to conect different providers to the same card. |
| 12:07.41 | hurdman | i'll try that |
| 12:07.50 | hurdman | i have to card on that server |
| 12:08.03 | hurdman | but i have a clock cable between my two card |
| 12:08.07 | kaldemar | WIMPy: what i meant was that it isn't the fault of the digium card or its driver alone. i've never used sanogma cards myself. |
| 12:08.30 | WIMPy | Neither have I. |
| 12:09.14 | WIMPy | hurdman: If you have enough ports to split the providers to different cards, that should solve your issue, but only if you remove the timing cable. |
| 12:09.28 | WIMPy | I think it can be disable by the driver. |
| 12:19.46 | *** join/#asterisk Vince-0 (~AndChat@196.215.188.241) |
| 12:24.01 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 12:26.09 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:28.16 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 12:30.06 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 12:31.12 | bbourdage | I am getting Unable to open DAHDI pseudo device, I have installed the dhadi package, any special config changes ? |
| 12:35.06 | ChrisInSydney | bbourdage are you referring to DAHDI dummy ? |
| 12:35.23 | bbourdage | I think so, I am a little confused on this. I am trying to get a meetme running |
| 12:35.29 | ChrisInSydney | K |
| 12:35.53 | bbourdage | I read a little about Dahdi Dummy, is that just an empty text file in the modules directory ? |
| 12:36.01 | ChrisInSydney | so, DAHDI is kernel dependent. |
| 12:36.20 | ChrisInSydney | if you have upgraded your kernel since compilling DAHDI, it probably wont work |
| 12:36.25 | bbourdage | Ok, I did install the apt-get asterisk-dahdi |
| 12:36.50 | ChrisInSydney | if you didnt have your kernel souce "stuff" in the right place, DAHDI wont compile either |
| 12:37.06 | ChrisInSydney | download the dahdi complete |
| 12:37.14 | bbourdage | I did from package, not source ? |
| 12:37.14 | ChrisInSydney | unpack to /usr/src |
| 12:37.45 | ChrisInSydney | build from souce, as sooon as the kernel rev changes, your DAHDI stops |
| 12:37.53 | ChrisInSydney | wget |
| 12:37.59 | ChrisInSydney | unpack to /usr/src |
| 12:38.07 | ChrisInSydney | make; make install |
| 12:38.15 | ChrisInSydney | make config |
| 12:38.36 | ChrisInSydney | that shouls do it. Check that the make runs |
| 12:38.59 | ChrisInSydney | as I said, if you dont have the kernel source bits it needs, it wont compile |
| 12:39.49 | bbourdage | Ok, Thanks I will try that |
| 12:40.15 | bbourdage | do I have to do any special conf changes ? |
| 12:40.55 | ChrisInSydney | Got a question for the group: Which setting in sip.conf needs to be changed to get rid of the word "asterisk" in the SIP chatter |
| 12:40.55 | kaldemar | bbourdage: asterisk-dahdi is just the userspace channel driver for asterisk. you need the kernel module that provides the timing functionality. apt-get install dahdi dahdi-linux |
| 12:41.15 | bbourdage | Kaidemar, I jsut tried to install that, and it says already installed |
| 12:42.02 | bbourdage | /usr/lib/asterisk/modules# apt-get install dahdi-linux |
| 12:42.02 | bbourdage | Reading package lists... Done,Building dependency tree,Reading state information... Done,dahdi-linux is already the newest version.,0 upgraded, 0 newly installed, 0 to remove and 14 not upgraded. |
| 12:42.14 | kaldemar | then the module is probably not loaded. confirm that with "lsmod | grep dahdi". if it doesn't show up, "modprobe dahdi". |
| 12:42.45 | ChrisInSydney | bbourdage: No special conf changes, it will just install over the packaged version's files |
| 12:43.19 | bbourdage | It does not show up the lsmod, when I try the modprob says module dahdi not found ? |
| 12:43.52 | kaldemar | bbourdage: and the "dahdi" package is installed aswell? |
| 12:43.53 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:45.13 | bbourdage | apt-get install dahdi,Reading package lists... Done,Building dependency tree,Reading state information... Done,dahdi is already the newest version. |
| 12:45.49 | ChrisInSydney | bbourdage: What happens when you stop and start the dahdi daemon ?? |
| 12:46.38 | bbourdage | I just rebooted, it will take a bit, learned that from working on windows :) |
| 12:47.29 | ChrisInSydney | bbourdage: /etc/init.d/dahdi stop; /etc/init.d/dahdi start |
| 12:47.34 | kaldemar | there is no such thing as dahdi daemon. |
| 12:47.46 | ChrisInSydney | Learned that in spite of windows ;-) |
| 12:47.51 | bbourdage | dahdi/system.conf not found |
| 12:48.28 | ChrisInSydney | kaldemar: You're technically correct |
| 12:48.45 | ChrisInSydney | What I meant to say was dahdi thingy |
| 12:49.15 | bbourdage | I am getting a not found, |
| 12:49.38 | ChrisInSydney | sounds like a kernel version issue |
| 12:49.50 | ChrisInSydney | build a new one |
| 12:49.57 | ChrisInSydney | dahdi, not kernel |
| 12:50.01 | kaldemar | bbourdage: what distro are you running? |
| 12:50.02 | ChrisInSydney | :o |
| 12:50.10 | bbourdage | debian squeeze |
| 12:50.24 | bbourdage | dahdi file is in the init directory |
| 12:52.12 | kaldemar | bbourdage: find /lib/modules/`uname -r`/ -name dahdi.ko |
| 12:52.30 | *** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:52.43 | kaldemar | bbourdage: that should list you a dahdi.ko with full path if you have it installed. |
| 12:52.53 | bbourdage | it is not on the system |
| 12:52.58 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:53.22 | kaldemar | oh well, do an "apt-get install --reinstall dahdi" and see if it gets in your system. |
| 12:54.08 | bbourdage | I just did a remove and then an install, but I will try the re-install |
| 12:54.32 | bbourdage | /etc/dahdi/system.conf not found. Nothing to do. |
| 12:55.48 | v0lZy | kaldemar: I checked with my ITSP. They support me setting the caller ID. Thing is that for some reason, this doesnt seem to work. note, i do have to set my caller id as name surname <number> (yes, with <> included) according to my interface. |
| 12:58.46 | ChrisInSydney | bbourdage: Failling that, maybe just build the dahdi "thingy" from source |
| 12:59.10 | bbourdage | I think you are correct, I am doing a little more searching, it seems to be a kernal issue with the package |
| 13:00.25 | ChrisInSydney | in my limited experience getting dahdi to work on CentOS, it was always easier to build from source |
| 13:00.33 | *** join/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 13:00.49 | catphish | are callgroups / pickupgroups still limited to 64 groups? |
| 13:01.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:01.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:03.17 | bbourdage | Would it be better to mess with the Daddi dummy ? |
| 13:04.37 | kaldemar | v0lZy: show what you have. |
| 13:05.11 | ChrisInSydney | bbourdage: In what way ? |
| 13:05.33 | kaldemar | catphish: yes |
| 13:05.50 | ChrisInSydney | Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from my.wan.ip.adr:5060 |
| 13:05.55 | bbourdage | I found some articles about using dahdi dummy, just by creating a file in the modules directory |
| 13:06.17 | v0lZy | kaldemar: I solved it. had to disable sending of from field |
| 13:06.35 | ChrisInSydney | any clues anyone ? Handset is using OpenSER on router as proxy |
| 13:06.46 | kaldemar | bbourdage: dahdi_dummy is the functionality that you need. looks like squeeze has an older version of dahdi that still has a separate module for dahdi_dummy. in newer versions of DAHDI the dummy functionality is built in the core module called "dahdi". |
| 13:06.57 | ChrisInSydney | issue is calling the handset |
| 13:07.02 | ChrisInSydney | from WAN |
| 13:07.32 | bbourdage | So short answer is, remove package and compile ? |
| 13:08.04 | ChrisInSydney | yes |
| 13:08.41 | ChrisInSydney | bbourdage: You will need the kernel source headers to compile against |
| 13:08.48 | kaldemar | or ask someone at #debian to help you install the package properly. |
| 13:09.10 | bbourdage | Kaldemar, great Idea, thank you |
| 13:09.37 | ChrisInSydney | kaldemar: Didnt they refer bbourdage to here ? |
| 13:10.16 | bbourdage | No, I have only been in this chat, just trying to get a page function to work |
| 13:12.02 | ChrisInSydney | bbourdage: I was being a little sarcastic. Give them a go. Failing that, you will get the result you are after by compiling from source. You can keep the packaged asterisk binaries. DAHDI can be a bitch |
| 13:12.21 | ChrisInSydney | good luck |
| 13:12.45 | leifmadsen | bbourdage: btw, there is no 'dahdi dummy' anymore -- it's just built into dahdi itself |
| 13:12.57 | bbourdage | Thanks, I did not catch on, had a couple thing running at once. I am already working on source compile. |
| 13:13.01 | leifmadsen | so just having dahdi loaded, with no modules, will give you kernel timing |
| 13:13.27 | bbourdage | Thanks Leif, I have not been able to get dahdi to load, but I have been using the packages |
| 13:13.33 | leifmadsen | odd... |
| 13:13.41 | leifmadsen | the ones from digium? |
| 13:13.46 | leifmadsen | might check with pabelanger who packages them |
| 13:13.59 | leifmadsen | I'd made sure you're using the digium repo, not something else |
| 13:14.12 | leifmadsen | I can't speak for the quality of the packages not built by pabelanger :) |
| 13:14.36 | catphish | kaldemar: is there a way to work around the limitation? |
| 13:14.56 | bbourdage | I have been just doing an apt-get , I will ask pabelanger when he comes on, thank you |
| 13:15.01 | leifmadsen | bbourdage: from here in particular: http://packages.asterisk.org/deb/ |
| 13:15.12 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:18.11 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 13:19.20 | *** join/#asterisk ChrisInSydneyToo (~Chris@60-242-81-231.tpgi.com.au) |
| 13:19.32 | kaldemar | catphish: change the source. |
| 13:20.30 | catphish | kaldemar: is it that simple? why would the limit exist? |
| 13:20.57 | catphish | looks like i might be better using Pickup() with either a wildcard or a PICKUPMARK |
| 13:23.12 | ChrisInSydneyToo | catphish: I patched the code in the pickup app to allow a variable other than PICKUPMARK |
| 13:23.42 | catphish | ChrisInSydneyToo: i'm happy using PICKUPMARK, I think that will work for me |
| 13:23.57 | catphish | it looks like its still context sensitive |
| 13:24.01 | ChrisInSydneyToo | the Variable starts with PICKUPMARK and then I would add the line and or extension after it and that allowed me to deal with multiple calls |
| 13:24.43 | ChrisInSydneyToo | if you have one number calling in thats ok, but if you need to separate out departments or tenants, I needed something more "selective" |
| 13:26.58 | ChrisInSydneyToo | Anyone seen this one: Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from my.wan.ip.adr:5060 |
| 13:27.49 | ChrisInSydneyToo | catphish: PICKUPMARK will work across contexts |
| 13:27.54 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 13:28.07 | catphish | ChrisInSydneyToo: i am separating tenants, but i don't really need selective pickup |
| 13:28.32 | catphish | so i should be able to use the tenant's ID as the PICKUPMARK on all inbound calls |
| 13:28.43 | ChrisInSydney | doesnt work like that :-( |
| 13:28.56 | ChrisInSydney | the word PICKUPMARK is the pickup mark |
| 13:29.02 | catphish | err |
| 13:29.04 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 13:29.06 | ChrisInSydney | Tenant 1 sets PICKUPMARK to 1 |
| 13:29.37 | ChrisInSydney | call comes in on tenant 2's number. PICKUPMARK is now set to 2. 1 can not pickup |
| 13:29.45 | catphish | so 1 is the pickup mark |
| 13:29.54 | ChrisInSydney | yup |
| 13:30.01 | catphish | that's exactly what i want |
| 13:30.33 | bulkorok | hi... is there a config file for res_fax_spandsp!? |
| 13:30.34 | ChrisInSydney | Not quite |
| 13:30.37 | ChrisInSydney | PICKUPMARK = 1 |
| 13:30.40 | catphish | then i just set tenant1's *8 to Pickup(1@PICKUPMARK) |
| 13:30.54 | ChrisInSydney | Dont think it works like that |
| 13:31.01 | catphish | how do you think it works? |
| 13:31.13 | ChrisInSydney | I'm just grabbing my dialplan |
| 13:31.18 | catphish | thanks |
| 13:31.23 | ChrisInSydney | what version are you running |
| 13:31.25 | ChrisInSydney | ? |
| 13:31.52 | catphish | 1.8.7.1 |
| 13:31.59 | ChrisInSydney | K |
| 13:32.47 | ChrisInSydney | ahh, now I see |
| 13:32.59 | ChrisInSydney | PickupChan() |
| 13:33.06 | bulkorok | just again: is there a config file for res_fax_spandsp!? |
| 13:33.28 | catphish | bulkorok: what are you trying to configure? |
| 13:33.37 | catphish | i believe the quick answer is no |
| 13:34.15 | bulkorok | I think so too, but I have to configure: T38FaxMaxDatagram, T38MaxBitRate |
| 13:34.50 | catphish | i think it goes in sip.conf [general] |
| 13:35.08 | bulkorok | yeah maxdatagram, but it wont be used ^^ |
| 13:35.28 | ChrisInSydney | catphish: core show application PickupChan |
| 13:36.02 | ChrisInSydney | not too sure if I patched the app tho |
| 13:36.15 | catphish | looks like pickupchan needs to be directed |
| 13:36.47 | catphish | i want an undirected (but scoped) pickup |
| 13:37.43 | ChrisInSydney | ahh. I changed it to take a partial string. So long as the channel "starts" with the string, you can do a generic pickup based on the tenant / department context |
| 13:38.08 | catphish | i see |
| 13:38.17 | catphish | well i'm going to give PICKUPMARK a shot |
| 13:38.21 | catphish | and see how that goes |
| 13:38.58 | ChrisInSydney | so if your extensions are tenant+extension, you could do a directed pickup on just the tenant portion of the call |
| 13:39.28 | ChrisInSydney | catphish: it will end in tears, been there and done that with 1.4 |
| 13:39.44 | ChrisInSydney | did you want the patch. I can DCC if I can find it |
| 13:41.06 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 13:41.08 | catphish | ChrisInSydney: go ahead and send me the patch |
| 13:41.26 | catphish | patching asterisk is a much bigger pain in the ass for me than changing the dialplan though |
| 13:42.07 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:44.43 | *** join/#asterisk bongfrog (~winston@c-50-134-135-61.hsd1.co.comcast.net) |
| 13:51.59 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:52.00 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 13:53.24 | *** join/#asterisk coppice (~coppice@m121-202-3-252.smartone.com) |
| 13:53.35 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:54.56 | *** join/#asterisk ccesario (~ccesario@177.35.100.170) |
| 13:55.30 | *** join/#asterisk serafie (~erin@nat/digium/x-mumrdiumrmvwhdaa) |
| 13:56.32 | ChrisInSydney | Anyone seen this one: Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from my.wan.ip.adr:5060 |
| 13:56.37 | ChrisInSydney | ?? |
| 13:56.53 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:57.13 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:58.46 | catphish | ChrisInSydney: what sent the 500? |
| 13:59.11 | wdoekes2 | ChrisInSydney: kamailio/error.c: err2reason_phrase with unknown error |
| 13:59.36 | ChrisInSydney | OpenSER on a router acting as a proxy for a handset behind NAT |
| 13:59.52 | ChrisInSydney | Milkfish on DDWRT |
| 14:00.51 | ChrisInSydney | works OK on 1.4. 1.8 it seems to choke |
| 14:00.55 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 14:02.00 | wdoekes2 | are you calling sl_reply_error before there is any error? |
| 14:02.14 | ChrisInSydney | wdoekes2: Broken xSER ?? |
| 14:02.56 | ChrisInSydney | hmm, I'll have to look at the cfg. Its just the default transparent proxy |
| 14:03.18 | ChrisInSydney | wdoekes2: BTW, Thanks |
| 14:03.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:03.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:03.35 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 14:03.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-vyxqeizgbvbqtvfu) |
| 14:03.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:07.10 | retentiveboy | Where should issues with the RPMs at packages.asterisk.org be reported? |
| 14:09.11 | wdoekes2 | to mjordan ;P |
| 14:09.18 | ChrisInSydney | wdoekes2: http://pastebin.com/LtAhpx5M |
| 14:09.30 | phix | kaldemar: sip to header |
| 14:09.50 | phix | can I escape it? |
| 14:10.40 | wdoekes2 | ChrisInSydney: add one of those log() statements before the sl_send_reply to see if you reach that |
| 14:10.53 | wdoekes2 | s/sl_send_reply/sl_reply_error/ |
| 14:11.40 | ChrisInSydney | Cool. I'll give it a go. Being on the router and auto generated, it may be a little bit of a challenge |
| 14:12.06 | ChrisInSydney | now the phone has "gone to sleep" |
| 14:12.17 | ChrisInSydney | probably what I should do |
| 14:12.22 | ChrisInSydney | 1:00am here |
| 14:13.07 | coppice | interesting. So Chris in Sydney doesn't refer to what I thought it might :-\ |
| 14:14.33 | ChrisInSydney | coppice: Hmmm now I am not sure whether to be curious or offended :P |
| 14:16.46 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 14:20.57 | ChrisInSydney | wdoekes2: Thanks for the pointers. The reg through the Ast 1.4 system works. This doesnt :-/ |
| 14:24.33 | ChrisInSydney | wdoekes2: Looks like a SIP header I am adding |
| 14:25.00 | ChrisInSydney | did a direct dial and it works, not through the extensions dial plan I have built |
| 14:25.04 | mjordan | retentiveboy: what's the issue you're having with the packages? |
| 14:25.37 | kaldemar | phix: you could CUT it first to get the part you want. |
| 14:26.22 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:29.17 | retentiveboy | mjordan: figured it out. seems a number of the asterisk18-* packages were automatically replaces with asterisk-* but not addons-core or addons-mysql. I manually removed those last two asterisk18-addon* packages and installed the asterisk-addon versions and I'm good. |
| 14:30.09 | ChrisInSydney | wdoekes2: CALLERID(NUM) is screwing things up |
| 14:30.39 | ChrisInSydney | Name is OK. num isnt |
| 14:31.19 | *** join/#asterisk k3asd` (~k3asd@static-94-32-127-180.clienti.tiscali.it) |
| 14:31.20 | ChrisInSydney | I'll look into it after some sleep |
| 14:36.04 | AdvoWork | is there a way with asterisk to check call quality issues? |
| 14:44.33 | *** part/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 14:45.57 | AdvoWork | anyone know much about jitterbuffer? |
| 14:47.04 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:47.06 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 14:48.31 | phix | kaldemar: how? |
| 14:49.37 | kaldemar | phix: core show function CUT |
| 14:52.14 | ChrisInSydney | good night all |
| 14:52.45 | kaldemar | phix: or func REPLACE |
| 14:56.19 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:56.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:58.33 | kaldemar | phix: $["${REPLACE(SIP_HEADER(To),\")}" =~ "5551234567"] |
| 15:03.41 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:09.06 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 15:17.12 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:19.24 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:25.22 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:31.45 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 15:32.35 | edge | Am i missing a easier way to call a group of people, i'm currently making these LONG lists with & |
| 15:33.25 | kaldemar | edge: queues |
| 15:34.37 | edge | kaldemar: can a call queue allow for staggered ringing? Like I want our basic employees to get the first 2 rings then the next level of managers and the basic employees for the next 4 then the last rings everybody with a phone |
| 15:35.54 | bulkorok | edge: you can configure a timeout for the queue and then call the next queue ins´the dialplan |
| 15:37.23 | bulkorok | edge: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 15:37.26 | edge | bulkorok: Where can i find documentation on that |
| 15:37.29 | edge | well you beat me to my question |
| 15:37.35 | bulkorok | :) |
| 15:37.48 | bulkorok | and for sure look at queues.conf |
| 15:37.50 | edge | I'm finding the documentation to be a bit difficult to find in a central way |
| 15:37.58 | bulkorok | ~book |
| 15:37.58 | 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 |
| 15:38.15 | edge | bulkorok: read the book. It was great, and i constantly go back to it. |
| 15:39.25 | edge | bulkorok: kind of skipped over chapter 13 thinking it was for like call centers, that may have been a mistake |
| 15:40.05 | bulkorok | yeah... queues are often used for callcenter things... but at least they are just a "group of channels" |
| 15:41.32 | edge | I also had a bit of trouble finding documentation about DAHDI in my dial plan. I'd like to pick lines for out going. So i'm assuming i have to check the status of the DAHDI port and check down the list until i find one open and usable. but couldn't find that syntax |
| 15:41.39 | kaldemar | don't use the voip-info page. in asterisk's CLI, "core show application Queue" shows dialplan usage for your version of asterisk and http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ACD.html is a good read. |
| 15:41.43 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nxvqhrasjybtrysd) |
| 15:41.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:43.26 | kaldemar | edge: define groups in chan_dahdi.conf and dial them with Dial(DAHDI/g0/<number>) for example. this would dial using all channels that are defined below group=0 and above another group definition in the config file. |
| 15:45.25 | edge | kaldemar: Can i go out of order? For example define InsuranceLines in the chan_dadhi.conf as being DAHDI port 1, 3, and 5 ? |
| 15:47.31 | kaldemar | edge: the groups are numbered, not named. and yes, you can order them how you like. |
| 15:49.07 | edge | kaldemar: I see the groups are what g0 , g1 mean. g3 being group 3. Is there a document with syntax i could read over? |
| 15:51.38 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup |
| 15:52.57 | kaldemar | see "logical groups" in http://svn.digium.com/svn/asterisk/tags/10.1.2/configs/chan_dahdi.conf.sample |
| 15:53.00 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 15:54.03 | edge | kaldemar: Thank you Very much. It has been a little challanging for me to phrase the question in such a way that google will show me what i'm looking for. |
| 15:54.30 | edge | kaldemar: will the second link work, even though i'm using the 1.8 stable branch |
| 15:55.44 | kaldemar | the group parameters have been the same since pre 1.0 versions, it will work. you can always change the tag version (10.1.2) to what you use. or just read the same sample file from your source package. |
| 16:00.17 | ChrisInSydney | wdoekes2: Fixed it. I had " characters that I no longer needed in some variable assignments. That broke the SIP addressing. All working now |
| 16:00.29 | ChrisInSydney | 3am I am off to bed |
| 16:00.36 | ChrisInSydney | night all |
| 16:02.11 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 16:03.58 | edge | kaldemar: If i understand this correctly, i create another [channels] block, define the channels that reside in it, and then assign a group number. What if they overlap? |
| 16:07.54 | kaldemar | edge: no, there can only be one [channels] block. chan_dahdi is a bit different from other config files. |
| 16:09.00 | kaldemar | edge: when you put some settings in the file, they apply to all channels that are listed with a "channel =>" line below them, unless overridden by another setting below. |
| 16:09.43 | kaldemar | edge: a channel can be assigned to more than one group. |
| 16:10.36 | wdoekes2 | good ChrisInSydney |
| 16:11.06 | wdoekes2 | those were custom headers? or do you have pedantic=no? |
| 16:11.09 | edge | kaldemar: so i'm looking at this config wrong. I should be saying under the same [channels] block listing channel => 1-2 and then below it assigning a group, then listing another channel => 2 - 3 then another group below it ? |
| 16:12.48 | kaldemar | edge: channels are defined only once. |
| 16:13.14 | kaldemar | group is not something that forms a configured entity, it is just a setting like any other. |
| 16:14.53 | kaldemar | if you want a channel to belong to two groups (0 and 1), you'd put group=0,1 above it. |
| 16:14.58 | edge | kaldemar: how do i then split a 8 port dahdi card into saying ports 1,3, and 5 are my insurance group and the rest (but also including 1 and 3 that we share and excluding 5)? |
| 16:17.14 | *** join/#asterisk zkn (~ergo@82.131.70.188.cable.starman.ee) |
| 16:17.21 | kaldemar | edge: http://pastebin.com/9qKXFBrQ |
| 16:18.19 | kaldemar | i see i mixed "=" and "=>", it doesn't really matter what you use, but usually "=>" is used for channels and "=" for others. |
| 16:18.21 | edge | kaldemar: Ok, i had a small understand, but i wasn't assigning 2 groups wat once to over lap |
| 16:19.34 | edge | I was also making the channel the first item, and the group a property of the channel |
| 16:20.06 | edge | Do i then assign context between group and channel and it will apply to those channels only? |
| 16:21.08 | rossand | Under sip show registry, I have a host that shows "No Authentication" state. Is there a command I can use to tell it to re-register? |
| 16:22.11 | rossand | sip show peers shows the host is OK. |
| 16:22.15 | kaldemar | edge: groups are just settings themselves. all settings apply to channels. context is for incoming calls, and groups have nothing to do with those. |
| 16:22.34 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 16:22.35 | edge | kaldemar: so this is wrong? er channel => 2 - 3 then another group below it ? |
| 16:22.58 | edge | kaldemar: http://pastebin.com/c6ng796L |
| 16:23.40 | rossand | does a sip reload disconnect currently active calls? |
| 16:23.42 | kaldemar | edge: nothing wrong there. |
| 16:23.54 | Qwell | rossand: no |
| 16:24.03 | rossand | Qwell: thank you very much. |
| 16:24.05 | edge | kaldemar: Great, i think i understand how that works then. You've been a great help |
| 16:24.15 | kaldemar | edge: no problem. |
| 16:24.26 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 16:25.07 | edge | kaldemar: some of the other examples show me setting things like signalling and usercallerid and such. If i set those and then don't change them they apply to ALL channels below that line , so long as i don't redefine it? |
| 16:25.20 | rossand | Qwell: did the trick. Thanks again |
| 16:26.15 | kaldemar | edge: yes. |
| 16:30.11 | *** join/#asterisk zkn (~ergo@82.131.70.188.cable.starman.ee) |
| 16:30.38 | edge | kaldemar: on the last line in group 1, 2-4 overlaps the defined channel 1,3 (the 3) is that ok? I redid it like this. I think it will work. http://pastebin.com/FLcBamZS |
| 16:34.29 | *** join/#asterisk af_ (~getsmart@78.134.18.236) |
| 16:41.25 | [TK]D-Fender | edge, Looks fine |
| 16:42.11 | edge | kaldemar: [TK]D-Fender I'm really glad you guys are so willing to help out. |
| 16:43.31 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:48.17 | *** part/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 16:50.53 | *** join/#asterisk gavimobile (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 16:53.21 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 17:05.24 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:30.16 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 17:30.43 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 17:35.22 | *** join/#asterisk joel_oliveira (~chatzilla@estrela-adm.nortenet.pt) |
| 17:37.12 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:38.15 | *** join/#asterisk jo3sm1th (~jo3sm1th@66.0.165.57) |
| 17:39.09 | jo3sm1th | what do you put in extensions.conf after exten => 12155592209,1,Answer() to make it ring to your SIP Phone (x lite) ? |
| 17:39.52 | *** join/#asterisk BitBandit (~taggmcd@rmatv.fttp.xmission.com) |
| 17:40.01 | WIMPy | a Dial |
| 17:40.15 | WIMPy | hurdman: Did you get on? |
| 17:40.25 | *** join/#asterisk arnotixe (~arnotixe@190.131.183.143) |
| 17:40.41 | jo3sm1th | o exten => 12137992209,2,Dial() |
| 17:40.46 | jo3sm1th | but what else, like in the parenthesis |
| 17:41.03 | WIMPy | 'core show application Dial' |
| 17:41.07 | arnotixe | hi all can I see DTMF coming or going at the asterisk server console somehow? |
| 17:41.12 | *** join/#asterisk TobSnyder (~schneider@146-52-59-49-dynip.superkabel.de) |
| 17:41.44 | WIMPy | Do you already have a pper for your phone? |
| 17:41.49 | *** part/#asterisk TobSnyder (~schneider@146-52-59-49-dynip.superkabel.de) |
| 17:41.53 | [TK]D-Fender | jo3sm1th, Dial(SIP/phone,30) <- ring device [phone] in sip.conf for 30 seconds |
| 17:43.03 | jo3sm1th | ok but do I actually put "phone" or the same name for hte phone that is sip.conf |
| 17:43.14 | jo3sm1th | because I change my username for the phone frequently, it is the only way to set the CID |
| 17:43.18 | [TK]D-Fender | jothe name of the peer you created in sip.conf |
| 17:49.50 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 17:50.16 | *** part/#asterisk mpe (~mpe@94.127.49.1) |
| 17:50.32 | joel_oliveira | Hello all! I have a PBX connected via PRI to an Asterisk 1.6.2.18 with an installed Digium Wildcard T1/E1 . The Asterisk itself connects via a SIP trunk to a Proxy-SIP. My problem is that I do not have control over the PBX and when I place outbound calls, they ring back to the user, but are being ended after 30 seconds. This happens when I get this output on Asterisk: "chan_dahdi.c: --... |
| 17:50.33 | joel_oliveira | ...Channel 0/28, span 1 got hangup request, cause 102". |
| 17:50.35 | joel_oliveira | As I understand the PBX decided to end the call because it has timed-out after 30 seconds and sent the ISDN code to the Asterisk. Am I understanding this correctly or am I wrong in my conclusion? |
| 17:53.24 | joel_oliveira | oh... I forgot to say: the calls weren't answered and this occurs on unanswered calls |
| 17:53.34 | WIMPy | Timeout, yes, but not on the call. |
| 17:54.13 | WIMPy | That is some sort of protocol error. |
| 17:56.09 | joel_oliveira | WIMPy: I see, where can I get more info on what's happening. I need to know if it's something wrong on the Asterisk machine or on the PBX so I can give more details to the responsible for the PBX. Thank you for your response |
| 17:56.51 | WIMPy | looks like they have some sort of trouble, understanding each other. |
| 17:57.03 | WIMPy | You need to look at pri debug to find out more. |
| 17:58.20 | joel_oliveira | WIMPy: via "pri set debug on span 1" right? |
| 17:59.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:59.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:59.26 | joel_oliveira | well, I'll look into that. Thanks for your time WIMPy |
| 18:03.58 | ajg | okay.. I've followed The Book exactly on configuring this. I'm still unable to get a softphone to register with my 1.8 server unless the SIP account is set to the extension |
| 18:04.16 | ajg | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html following this guide |
| 18:05.42 | ajg | any time I try and use any sip account name other than the extension, I get No matching peer found |
| 18:06.34 | *** join/#asterisk flan (~flan@206.174.214.3) |
| 18:07.21 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:08.03 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 18:09.43 | flan | Does anyone know if there's a config option to get Asterisk to log the AMI messages it's emitting? |
| 18:13.35 | flan | Alternatively, has anyone experienced problems with the "ConfbridgeEnd" message suddenly disappearing (non-intermittent) without any known changes to an environment? |
| 18:18.38 | *** join/#asterisk scgm11 (~Sebastian@r186-50-67-162.dialup.adsl.anteldata.net.uy) |
| 18:22.42 | ajg | whee |
| 18:22.50 | ajg | appears to have been a C3X problem |
| 18:22.54 | ajg | zoiper works fine |
| 18:25.51 | *** join/#asterisk jamicque (~jamicque@80.50.125.74) |
| 18:26.26 | jamicque | hi is there any way to autoload realitme peers during chan_sip reload? |
| 18:27.35 | hurdman | WIMPy: what's up ? |
| 18:27.38 | *** join/#asterisk jo3sm1th (~jo3sm1th@66.0.165.57) |
| 18:29.29 | *** join/#asterisk nwykes (~nwykes@184-96-110-206.hlrn.qwest.net) |
| 18:31.07 | jo3sm1th | I have an extension in sip.conf [13105551212] and in order to register a softphone also it has that as the username/authorization name in it. So since I change these values frequently would it then be impossible to receive calls via DID because I am constantly changing |
| 18:32.01 | jo3sm1th | the name of the extensions in sip.conf , and the register name in Xlite. Since you got to configure inbound calls in extensisons.conf to Dial the Sip phone at 13105551212 or whatever, since my sip phone username chnages sometimes then it will make routing DIDs to it impossible right |
| 18:32.04 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 18:32.06 | *** part/#asterisk nwykes (~nwykes@184-96-110-206.hlrn.qwest.net) |
| 18:33.17 | *** join/#asterisk thurin (~nwykes@184-96-110-206.hlrn.qwest.net) |
| 18:34.21 | Katty | weigh anchor and hoist the mizzen! tonight we be three sheets to the wind, our rum a fizzin!!! |
| 18:54.15 | flan | Okay, dismiss my problem as "unrelated buffer overflow", since I was doing something with a struct that might be illegal in some countries. |
| 18:55.26 | *** part/#asterisk scgm11 (~Sebastian@r186-50-67-162.dialup.adsl.anteldata.net.uy) |
| 18:55.57 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 19:12.15 | Kobaz | when i do sip show peer x |
| 19:12.15 | Kobaz | <PROTECTED> |
| 19:12.16 | Kobaz | is that the defaultuser value |
| 19:12.18 | [TK]D-Fender | jo3sm1th, why is the username changing all the time? |
| 19:12.18 | jo3sm1th | because i got like 10 different CIDs that I use |
| 19:12.18 | Kobaz | when i set defaultuser= in the sip peer, it doesn't get used |
| 19:12.30 | jo3sm1th | So am I right, as far as logically, I will have problems using DIDs to ring to the softphone unless I had 2 softphones open, one with a static username right |
| 19:13.59 | [TK]D-Fender | jo3sm1th, I think you have some misunderstandings about "callerid" and where this can and should apply |
| 19:14.13 | Kobaz | seems like a bug |
| 19:14.41 | [TK]D-Fender | jo3sm1th, Why is it "changing' al the time and who is supposed to see this change? |
| 19:17.19 | jo3sm1th | Ok... I duno what CID is, whatever |
| 19:23.00 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:25.13 | ChannelZ | your SIP peer names and your DIDs aren't really related. |
| 19:32.43 | sawgood | ChannelZ: you helped me the other day with a SIP port 5060 and cable modem concern (it turned out the ISP was for sure blocking UDP on ports 5060-5069) |
| 19:32.52 | sawgood | Thanks again for your time ... all is well now |
| 19:35.13 | p3nguin | I can't remember... did 1.4 have faxdetect? |
| 19:38.18 | cmendes0101 | I have a polycom ip phone that is not busy or on a call but when trying to call it says congested and goes to voicemail. Asterisk says no channels in us and phone says line is idle. Any ideas? |
| 19:39.00 | *** join/#asterisk Fubard (~brawr@vpn.bctconsulting.com) |
| 19:39.39 | ChannelZ | sawgood: glad you got it working. Hate @#*ing port-blocking ISPs |
| 19:40.02 | p3nguin | cmendes0101: Show us the sip debug. |
| 19:40.29 | Fubard | at&t u-verse in the states port blocks but does not admit to it >.< |
| 19:41.23 | bmoraca_work | has anyone tried running Asterisk on LVS? (linux virtual server) |
| 19:43.43 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 19:44.29 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 19:45.13 | cmendes0101 | p3nguin: http://pastebin.com/DmSgPTYM |
| 19:46.27 | p3nguin | SIP/2.0 404 Not Found |
| 19:46.41 | p3nguin | Perhaps you haven't configured asterisk to work with NAT. |
| 19:47.32 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 19:47.40 | p3nguin | Actually, that probably is not the problem in this case. |
| 19:47.42 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 19:48.33 | McBoingBo | I have a user where the phone keeps rebooting/crashing, once a week I would say...I changed: phone, network plug, network cable, and still crashing, could an extension be doing this? |
| 19:48.52 | *** join/#asterisk lcat (~lcat@187.45.253.206) |
| 19:49.30 | Fubard | McBoingBo: sounds like something on the network. is this on a PoE switch perhaps? |
| 19:49.41 | McBoingBo | Fubard, yeah POE switch |
| 19:50.05 | Fubard | are any other poe devices losing power? |
| 19:50.23 | McBoingBo | none |
| 19:50.39 | McBoingBo | only thing that is the same is their phone itself, I changed everything else |
| 19:50.57 | McBoingBo | but I havent tried to put it on AC power instead |
| 19:51.02 | Fubard | for shiggles, try switching the port on the switch that the device is connected to; pcould be the port losing power |
| 19:51.44 | McBoingBo | true, but changing the network plug location twice, it would be unlikely several ports are dying and they were always the same user, still possible I guess will keep in mind thanks |
| 19:51.54 | Fubard | ah, well definitely try switching the phone itself out first if you want to rule out the problem, if you are more concerned with just getting it fixed then move the cat5 over to a new port on the switch as well |
| 19:52.17 | Fubard | what is the phone model/make? |
| 19:52.46 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 19:53.17 | McBoingBo | Polycom 430 |
| 19:53.26 | McBoingBo | SoundPoint IP 430 I mean |
| 19:53.54 | cmendes0101 | p3nguin: It was working yesterday and the network cable was bad so replaced it today and now that keeps happening |
| 19:54.57 | Fubard | McBoingBo: could try using a poe patch cable (not sure what exactly its called, i've only seen polycom make these) |
| 19:55.38 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 19:59.13 | Fubard | McBoingBo: i don't mean to be stuck on it being a poe issue; its very possible its not. in my personal experience poe has been sketchy so i may have a degree of bias here. :) |
| 19:59.30 | Fubard | if you have a spare phone try swaping it out |
| 19:59.36 | McBoingBo | hehe, no worries I appreciate the suggestions/ideas |
| 20:00.09 | McBoingBo | Fubard, I did, I swapped everything except the extension itself, and now will try adapter instead of POE |
| 20:08.20 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:11.38 | *** join/#asterisk libryder (~Adium@67.204.184.34) |
| 20:12.06 | libryder | anyone have any ideas how to troubleshoot laggy dtmf on a production machine? the console moves way too fast |
| 20:12.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:12.41 | Fubard | is this an intuitive voice box? |
| 20:12.42 | libryder | i'm trying to figure out if the lag is on our system or the carrier |
| 20:12.47 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 20:13.08 | libryder | no |
| 20:13.15 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:13.23 | [sr] | hellou my friends |
| 20:13.27 | Fubard | are you using inband or out of band? |
| 20:13.56 | [sr] | i'm trying to do a test system to be more updates on how some stuff work, and i'm doing it all from source with time and paciente, for testing |
| 20:14.35 | libryder | Fubard: RCF2833 |
| 20:15.55 | [sr] | my 1st question is, for format interpretters, i already added the liboggvorbis-dev run configure & make menuconfig |
| 20:16.14 | [sr] | but the "format_ogg_vorbis" still not available? |
| 20:16.22 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:16.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:16.42 | [sr] | i just want to understand better how can i enable and see their deps (normaly -dev files on system?) |
| 20:23.28 | [sr] | for example, chan_sip is marked with XXX, which dev files are required? the h323 of sofia-dev ? |
| 20:28.17 | [sr] | oh i think i've got it how can i get the deps :) |
| 20:38.37 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 20:39.23 | *** join/#asterisk netman (netman@73.227.76.188.dynamic.jazztel.es) |
| 20:41.50 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 20:43.46 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 20:44.27 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 20:51.59 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 20:58.03 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 21:01.46 | sawgood | ChannelZ: Thanks for the netcat tips .. The ISP only took 2 business days to open the blocked ports |
| 21:02.04 | *** join/#asterisk darkbasic_ (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 21:04.27 | *** join/#asterisk libryder (~Adium@70.103.79.251) |
| 21:19.24 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 21:23.28 | ChrisInSydney | Mornin' all |
| 21:24.20 | ChrisInSydney | whats the setting in sip.conf to get rid of the asterisk@hostname.domain out of the sip headers ? |
| 21:25.05 | ChrisInSydney | done useragent, engine, sdpsession, mwi_from, but its still there in the SIP |
| 21:25.44 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-hskdcvhelqcogkjn) |
| 21:28.38 | *** join/#asterisk luckyaba (Lucky@ip68-6-93-108.sb.sd.cox.net) |
| 21:31.42 | *** join/#asterisk bmg505 (~leon@196-209-152-217.dynamic.isadsl.co.za) |
| 21:32.24 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
| 21:32.38 | ChrisInSydney | anyone ? I would like to change From: "asterisk" <sip:asterisk@myserver.mydomain> |
| 21:32.57 | ChrisInSydney | v1.8.9.x |
| 21:42.30 | woleium | Is there any way to ask asterisk what it's allows network are from the cli? |
| 21:44.07 | woleium | srry, so be more specific, which networks SIP clients are allowed to connect from? |
| 21:46.11 | woleium | I have a bunch of successfully connected clients on a local network, but one on a remote refuses to register. I can see it hitting tftproot for config files, but sip set debug ip [it's ip] shows nothing after a reboot :-( |
| 21:47.35 | leifmadsen | ChrisInSydney: change it to what? |
| 21:49.45 | ChrisInSydney | leifmadsen: Something else that doesnt say asterisk |
| 21:49.51 | ChrisInSydney | like pbx@ |
| 21:54.47 | *** join/#asterisk bmg505 (~leon@196-209-152-217.dynamic.isadsl.co.za) |
| 21:55.31 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
| 21:58.15 | *** join/#asterisk eagle00789 (~chatzilla@54190191.cm-5-2a.dynamic.ziggo.nl) |
| 21:58.47 | ChrisInSydney | leifmadsen: Is this possible from sip.conf or is it a "patch" to chan_sip.c ? |
| 21:58.51 | [sr] | people |
| 21:58.54 | eagle00789 | question. i have asterisk setup and can use it internally, but i want to dial someone @ bla@sip2sip.info how would i set that up |
| 21:59.06 | [sr] | the menuselect options are saved in any special file? like the .config from kernel |
| 21:59.31 | ChrisInSydney | eagle00789: Dial(SIP/bla@sip2sip.info) |
| 21:59.46 | eagle00789 | is that all?? |
| 21:59.46 | ChrisInSydney | you also need srvlookup=yes in sip.conf |
| 22:00.18 | ChrisInSydney | pretty much. You can test SIP/200901@login.zipdx.com |
| 22:00.52 | eagle00789 | the bla@sip2sip is from my mobile. will try that. thansk anyway |
| 22:00.58 | ChrisInSydney | VoIP Users Conference. Not running but one to test. There is a list of SIP urls to test on voip-info |
| 22:01.31 | eagle00789 | ah. ok. thx |
| 22:01.40 | ChrisInSydney | if the handset doesnt support SIP URL dialling, you will have to do some dial plan things |
| 22:01.56 | ChrisInSydney | ie 882 = vuc |
| 22:02.22 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
| 22:02.33 | leifmadsen | ChrisInSydney: 'asterisk' is the username |
| 22:02.42 | leifmadsen | place a call using a different username |
| 22:02.57 | ChrisInSydney | hmm |
| 22:03.01 | ChrisInSydney | I'll have a look |
| 22:03.20 | ChrisInSydney | using Snoms with 8.4.32 |
| 22:03.23 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:03.27 | ChrisInSydney | PBX type is set to asterisk |
| 22:03.32 | leifmadsen | ChrisInSydney: sounds like you're placing a call without setting up an account to call from |
| 22:04.14 | ChrisInSydney | leifmadsen: Extension to extension |
| 22:04.18 | ChrisInSydney | both registred |
| 22:04.22 | leifmadsen | ok... |
| 22:04.50 | ChrisInSydney | asterisk is not running under a system account asterisk |
| 22:05.06 | leifmadsen | that doesn't really have anyting to do with anything |
| 22:05.30 | leifmadsen | you'll have to pastebin more information then what you've provided |
| 22:06.50 | ChrisInSydney | Cool. I'll try and get some meaningful dumps. Thought it might be a simple parameter in sip.conf. such as fromuser=somethingelse |
| 22:07.35 | ChrisInSydney | leifmadsen: Thanks heaps, I have to go to do some "real" work now. after 9am here |
| 22:08.05 | ChrisInSydney | I'll try some things and pastebin what I find |
| 22:08.18 | ChrisInSydney | c yaz |
| 22:09.02 | Fubard | libryder: sorry for the delay. inband(rfc2833) means that there are flags in the packet header that tell the gateway to produce the dtmf on their end. first off you want to be sure your gateway supports this. secondly you want to be sure your phone is sending dtmf in rfc2833 format. pcap is your friend here. |
| 22:09.51 | Fubard | ... oh he's not even in here anymore lol |
| 22:11.17 | p3nguin | inband is not rfc2833. |
| 22:11.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:14.18 | *** join/#asterisk tmrhmdv (~subway@ridezap.com) |
| 22:14.39 | tmrhmdv | unidentified |
| 22:16.05 | *** part/#asterisk tmrhmdv (~subway@ridezap.com) |
| 22:22.54 | eagle00789 | ChrisInSydney: thx. it works like a charm :D |
| 22:28.26 | eagle00789 | each time i make a change to extensions.conf, do i have to restart asterisk? |
| 22:28.56 | l0st-soul | reload will usually suffice |
| 22:29.03 | eagle00789 | ok |
| 22:29.54 | l0st-soul | even dialplan reload |
| 22:31.20 | eagle00789 | core reload will not suffice |
| 22:31.24 | eagle00789 | ?? |
| 22:32.28 | l0st-soul | dialplan reload will reread extensions.conf |
| 22:32.39 | l0st-soul | reload will reread everything |
| 22:32.44 | l0st-soul | core reload i dont know |
| 22:33.08 | l0st-soul | at least on my 1.4 :) |
| 22:34.06 | eagle00789 | i'm running 1.8 |
| 22:34.44 | eagle00789 | core reload: Global reload |
| 22:35.19 | ajg | eagle00789: you want dialplan reload for extensions.conf changes |
| 22:37.41 | eagle00789 | i know that now. l0st-soul told me... |
| 22:37.58 | p3nguin | Anyone here know where to find VAD settings in a Linksys SPA-942? The ITSP claims this is the reason Asterisk bitches that their system is sending comfort noise and said to check it in the phones. |
| 22:38.41 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 22:39.55 | p3nguin | eagle00789: dialplan reload is the correct way to reload extensions.conf after changes are saved. |
| 22:40.11 | eagle00789 | i know that now |
| 22:40.17 | p3nguin | Great! |
| 22:40.32 | p3nguin | Now there won't be anymore senseless reloading of things that don't change. |
| 22:43.27 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 22:48.28 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 22:50.53 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:51.42 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:00.39 | *** part/#asterisk eagle00789 (~chatzilla@54190191.cm-5-2a.dynamic.ziggo.nl) |
| 23:10.41 | WIMPy | hurdman: I was just interested if you cold get it reliable by changing ports. |
| 23:16.48 | *** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 23:24.24 | *** join/#asterisk RickCogley_ (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 23:25.23 | *** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 23:27.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nxvqhrasjybtrysd) |
| 23:29.08 | *** join/#asterisk sp00kz (~ilubj00@green.thegeeks.us) |
| 23:32.23 | phix | ok so what would block voice? or have the appearence of blocking it? |
| 23:33.14 | phix | and what tests can I do? |
| 23:33.35 | phix | TO determine what is causing it? |
| 23:37.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:39.17 | *** join/#asterisk thurin (~nwykes@184-96-125-83.hlrn.qwest.net) |
| 23:42.15 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 23:45.57 | *** part/#asterisk mpe (~mpe@94.127.49.1) |
| 23:55.37 | phix | What config files should i paste up? |