| 00:13.52 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 00:34.56 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:09.08 | *** join/#asterisk syntaxman (wade@funtoo/user/syntaxman) |
| 01:14.50 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 01:38.52 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 01:47.18 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-ylthpbiksfhqhlok) |
| 01:53.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:54.48 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 01:55.36 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 02:00.31 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 02:09.01 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 02:09.31 | *** join/#asterisk yano (~yano@freenode/staff/yano) |
| 02:27.43 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 02:36.05 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-56-241.om.om.cox.net) |
| 02:46.17 | *** join/#asterisk CeBe (~CeBe@port-92-206-6-83.dynamic.qsc.de) |
| 02:53.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 03:05.51 | *** join/#asterisk am1n0 (~devnull@unaffiliated/am1n0) |
| 03:05.54 | am1n0 | hello |
| 03:08.13 | am1n0 | i'm having trouble with outbound calls with a cisco 7905g hardphone. it seems that the phone, to initiate a call, is sending two INVITEs, and asterisk is sending an OK to the first one, but the phone will only accept an OK for the second, retransmitted INVITE, so asterisk keeps sending the OK over and over, and ends up destroying the dialog. inbound calls work fine, presumably because they don't involve the phone re-transmitting INVITEs. |
| 03:08.26 | am1n0 | i've pastebinned my sip.conf and a sip debug log. let me link them here. |
| 03:09.43 | am1n0 | here is my sip.conf http://pastebin.com/raw.php?i=bWzegiSX |
| 03:09.44 | am1n0 | here is my sip debug http://pastebin.com/raw.php?i=XYACQv4E |
| 03:09.59 | am1n0 | for the record, my phone is at 10.0.69.6 and my asterisk box is sitting at 10.0.69.100 |
| 03:18.23 | shido6 | what version of asterisk? |
| 03:20.12 | am1n0 | 11.9.0 |
| 03:20.12 | shido6 | 11.9.0 |
| 03:23.50 | am1n0 | i can get you a log from the phone itself, if you think that'd help. |
| 03:28.33 | [TK]D-Fender | maybe it'll show something |
| 03:28.56 | [TK]D-Fender | But that is a duplicate call it's trying to pass off... everything looks legit so far execpt for its behavior |
| 03:29.21 | [TK]D-Fender | You might try upgrading the firmware another step or two |
| 03:29.29 | [TK]D-Fender | Could be a glitchy version |
| 03:31.47 | am1n0 | i tried that, but this is as far as i can go, and it's EOL |
| 03:31.56 | am1n0 | and if you see here, http://pastebin.com/raw.php?i=cJPdrSvV |
| 03:32.19 | am1n0 | for whatever reason, after the cisco's INVITE, it's thinking that asterisk's Trying is a 'bad msg' |
| 03:32.25 | am1n0 | unfortunately, only cisco would know what that means |
| 03:32.27 | am1n0 | or why |
| 03:32.31 | [TK]D-Fender | I've seen 8.0.1 referenced.. may be a few other minor revisions for it |
| 03:32.54 | am1n0 | i'll search around. as a hack, there's no way to get asterisk to ignore the first INVITE? |
| 03:33.55 | am1n0 | or alternatively, can i use the SCCP firmware with asterisk? |
| 03:35.36 | shido6 | why is nat on |
| 03:35.49 | am1n0 | in sip.conf ? |
| 03:36.34 | shido6 | 10.0.69.6 <âis there some strangeness betweeen these two IPs?â> 10.0.69.100 ? |
| 03:36.48 | am1n0 | no, they're hooked up to the same router |
| 03:37.17 | shido6 | in sip.conf, yep - nat is on - why is that? |
| 03:37.58 | am1n0 | was just trying anything. nat=no has same results. |
| 03:40.46 | shido6 | nat, directmedia, and insecure - remove - reload then test inbound - work? try outbound. |
| 03:42.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 03:47.24 | am1n0 | removed them. inbound worked just fine, but outbound doesn't even ring the other extension. let me pastebin you the new sip debug |
| 03:49.12 | am1n0 | http://pastebin.com/raw.php?i=3MHG5UQ2 |
| 03:49.28 | am1n0 | i omitted them, but asterisk then just keeps retransmitting the unauthorized messages |
| 03:56.47 | am1n0 | i found firmware version 8.0.1. let me upgrade and see if that solves this problem. |
| 04:06.33 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 04:11.24 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 04:34.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:35.58 | am1n0 | ok, i upgraded to firmware version 8.0.1. no change, without the entries in sip.conf you said to take out, inbound works fine, but outbound doesn't work at all, with the 401 unauthorized error. |
| 04:36.41 | am1n0 | however if i put in insecure=invite back into sip.conf and call another extension and pick up, i can get a few seconds of audio back and forth before i get the congestion tone, so rtp is being set up... |
| 04:51.11 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
| 05:26.45 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 06:10.09 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 06:45.48 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 06:59.01 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 07:16.06 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 07:41.05 | shido6 | then treated like an attack? |
| 07:41.32 | shido6 | send a pcap |
| 07:42.08 | *** join/#asterisk Chinorro (~Chino@89.17.194.254) |
| 07:50.18 | shido6 | insecure=port,invite |
| 07:52.28 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 07:54.56 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:00.09 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 08:03.29 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:07.25 | *** join/#asterisk riess82 (~riessma@88-117-56-56.adsl.highway.telekom.at) |
| 08:12.56 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 08:30.41 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 08:43.17 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 08:59.05 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 09:08.00 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-6-83.dynamic.qsc.de) |
| 09:40.12 | *** join/#asterisk 77CAAA56S (~Thunderbi@cpc5-nott16-2-0-cust33.12-2.cable.virginm.net) |
| 09:43.56 | *** join/#asterisk puzzled (~patrick@54692558.cm-12-2a.dynamic.ziggo.nl) |
| 10:08.50 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-biypzjuiahcdacqq) |
| 10:27.57 | *** join/#asterisk af_ (~af@93-43-45-195.ip90.fastwebnet.it) |
| 10:53.20 | *** join/#asterisk BitEvil (~quassel@tor/regular/SpeedEvil) |
| 10:56.37 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 10:59.15 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 11:22.13 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 11:53.01 | *** join/#asterisk infobot (ibot@rikers.org) |
| 11:53.01 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 12:00.26 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:05.27 | *** join/#asterisk vedic (~V@183.82.105.37) |
| 12:05.34 | *** join/#asterisk p0g0 (~pogo@unaffiliated/p0g0) |
| 12:29.42 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:49.50 | vedic | Hey guys, one of my friend has got sales pitch from Enghouse Interactive saying that Asterisk is not good for large scale deployments and better go for their offerings (Avaya, NEC, CISCO etc). I would like to read some case study where somebody has done large deployment for TDM channels (IVR, Voice mail, conference bridge etc). Large means 2000+ ports (eg: a public bank or govt offices) |
| 12:51.22 | vedic | I am sure involves load balancers, cluster of servers, RAM disks or other cache etc. But is there any such case? May a call center for Bank or other big enterprise? |
| 12:55.29 | vedic | If you find anything relevant to my query, you can also drop me an email on developer.vm@gmail.com . My friend will be happy to talk to you. |
| 13:06.02 | WIMPy | That's an interesting argument. |
| 13:07.14 | WIMPy | But The key point is that you probably don't want to use TDM ports in the first place. |
| 13:08.07 | WIMPy | With Asterisk you use SIP phones. If that's a good idea by itself is another question. |
| 13:12.23 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 13:20.37 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:20.41 | *** join/#asterisk captain118 (ad15584b@gateway/web/freenode/ip.173.21.88.75) |
| 13:21.31 | captain118 | I've gotten myself in a inner loop can someone help me? |
| 13:21.56 | captain118 | when I dial out it dials the number I am calling from. |
| 13:22.56 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 13:23.50 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 13:27.21 | captain118 | fstd are you available to help me troubleshoot an issue? |
| 13:28.40 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 13:38.49 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 13:53.12 | fstd | captain118: sure! |
| 13:56.46 | *** join/#asterisk Torenn (~Valinor@mimas.lightwitch.org) |
| 13:58.06 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:13.20 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 14:18.35 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:28.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:37.26 | *** join/#asterisk vedic (~V@122.169.159.3) |
| 14:37.31 | vedic | WIMPy: Pls note the two ends of channels: User with Landline or GSM phone ---------Asterisk running IVR/Voice mail/Conference etc. Now could you pls elaborate on what you mean by TDM ports? |
| 14:37.56 | vedic | What you mean by don't need TDM ports? |
| 14:39.12 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 14:41.34 | puzzled | vedic: you don't necessarily need TDM ports (a T1 or E1) for trunks. You can also use SIP trunks |
| 14:49.43 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-ywhrkastgttfafen) |
| 14:51.01 | vedic | puzzled: You mean Teleco will provide SIP trunk? |
| 14:51.08 | puzzled | vedic: yes |
| 14:52.14 | vedic | puzzled: So it happens this way (correct me if wrong): Users <--PSTN/GSM--> telecom <--SIP trunk--> * server on the cloud. |
| 14:53.24 | puzzled | vedic: more like: users < - > * server on premise < - > SIP trunk < - > Telco |
| 14:53.52 | puzzled | I would never use a VoIP server "in the cloud" with 2000 ports |
| 14:54.49 | vedic | puzzled: How users will connect to server on premise? |
| 14:55.12 | vedic | Puzzled: users are on Landline or Mobile phones |
| 14:55.22 | puzzled | vedic: a user's SIP phone will use the LAN to connect via SIP to the * sever |
| 14:55.25 | puzzled | server |
| 14:55.48 | vedic | puzzled: User is using non IP phone. They are GSM or Landline (BRI) lines |
| 14:56.51 | puzzled | vedic: not sure I understand you. do you have a drawing? |
| 14:57.33 | [TK]D-Fender | vedic: So you have people at various points on the PSTN. What do you want them to do? |
| 14:58.29 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 15:09.27 | *** join/#asterisk roxlu (~roxlu@5469B62A.cm-12-2c.dynamic.ziggo.nl) |
| 15:17.14 | *** join/#asterisk ShapeShifter499 (~ShapeShif@unaffiliated/shapeshifter499) |
| 15:17.48 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 15:18.23 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 15:38.29 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 15:48.35 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 15:58.17 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 15:58.35 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 16:06.19 | *** join/#asterisk Alex_Bkash (cbdf5d16@gateway/web/freenode/ip.203.223.93.22) |
| 16:19.00 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 16:25.19 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:51.09 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 16:55.50 | *** join/#asterisk Alex_Bkash (cbdf5d16@gateway/web/freenode/ip.203.223.93.22) |
| 17:07.53 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 17:12.32 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 17:15.30 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 17:25.05 | *** join/#asterisk riess82 (~riessma@178-190-161-229.adsl.highway.telekom.at) |
| 17:39.19 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 17:51.54 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 17:55.36 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 18:00.26 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 18:04.46 | *** join/#asterisk Alex_Bkash (7c06ecfc@gateway/web/freenode/ip.124.6.236.252) |
| 18:08.33 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 18:14.43 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 18:16.36 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 18:21.19 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 18:43.29 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 18:56.48 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 19:01.34 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 19:02.00 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
| 19:02.59 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 19:04.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 19:05.58 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-mbozilzshtbpqwoe) |
| 19:07.12 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 19:12.42 | *** join/#asterisk igcewieling (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 19:14.43 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 19:29.12 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 19:48.42 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 20:01.36 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 20:03.13 | *** join/#asterisk captain118 (ad15584b@gateway/web/freenode/ip.173.21.88.75) |
| 20:20.40 | *** join/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 20:20.43 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 20:34.13 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 21:02.22 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 21:41.54 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
| 21:42.22 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 21:50.26 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 21:55.03 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 22:03.07 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 23:25.42 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 23:28.29 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 23:45.13 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 23:45.33 | dan_j | Hi, |
| 23:46.29 | WIMPy | lo |
| 23:46.35 | dan_j | My asterisk server keeps showing most of the peers unreachable, and then after a few seconds, they all come back. |
| 23:46.42 | dan_j | Happens around once every 10 minutes. |
| 23:47.09 | WIMPy | How many peers do you have? |
| 23:47.22 | dan_j | Is there a tool i can use to create a sort of brute force test, so i can see when asterisk stops responding |
| 23:47.39 | dan_j | or if its not asterisk, at least i can see where the packets are getting stuck. |
| 23:47.40 | WIMPy | sipsak |
| 23:47.55 | WIMPy | But it might be itself sending out the options requests. |
| 23:48.29 | dan_j | I assumed as much. At least i can test incoming connections first. |
| 23:48.51 | dan_j | outbound will be a bit harder. I'm running tcpdump to capture all the packets next time it happens. |
| 23:53.52 | *** part/#asterisk hecatae (~philip@host-92-28-13-14.as13285.net) |
| 23:57.38 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |