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| 02:31.06 | fubada | hi, i noticed my router is negotiating at 100mbit HALF with my * bx |
| 02:31.18 | fubada | does that mean anything bad for sip/rtp? |
| 02:40.06 | slav3_kitten | means you get 50mbit up an down |
| 02:40.11 | slav3_kitten | that's a ton of traffic |
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| 02:41.17 | slav3_kitten | so i'd not see anything really of issue with it in of itself |
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| 02:42.00 | slav3_kitten | but, fubada it might be alerting you to bad cabling, bad nick, or bad router port. i'd check it with appropriate test equipment |
| 02:43.24 | slav3_kitten | derp i had a derp moment, it's not 50mbit up/down it's 100 tx or 100 rx but both can't happen at the same time |
| 02:43.28 | fubada | thanks guys |
| 02:43.35 | fubada | its only a t1 pipee |
| 02:43.55 | slav3_kitten | an on that note i think i need to get some shut eye because i',m exhausted |
| 02:45.25 | fubada | gnight |
| 02:45.39 | slav3_kitten | night *collapses |
| 02:51.59 | jpsharp | If both ends negotiate 100/half, then its okay (not good, but okay). If one end negotiated 100/full and the other end 100/half, bad stuff happens. |
| 02:52.13 | jpsharp | YOu get all kinds of retries, collisions, and nastyness. |
| 02:53.15 | slav3_kitten | that too |
| 02:53.45 | slav3_kitten | jpsharp, remember back in the day lan parties with hubs because switches were too expensive? |
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| 02:55.17 | jpsharp | Yep. |
| 02:56.09 | slav3_kitten | collision light made me flash back to the good ol days |
| 02:56.29 | slav3_kitten | which then made me remember running lans on 10-base cards and thinnet |
| 02:57.02 | jpsharp | I used to work for a large telemarketing company that ran the agent workstations (1000 of them) across hubbed 10Mbit. |
| 02:58.02 | jpsharp | 120 stations per large hub, the hub was connected to an aggreating router via 10Mb thinnet. |
| 02:58.08 | slav3_kitten | so it didn't really flash as much as was on steady :P |
| 02:58.31 | jpsharp | Pretty much. All the agent stations ran very chatty NetBIOS over IPX. |
| 02:59.11 | slav3_kitten | ah the good ol days |
| 02:59.29 | slav3_kitten | when networks sucked balls but we didn't care because we didn't know better |
| 02:59.37 | jpsharp | The only thing that made the network stable was the aggregating routers were tied together via 100Mbps FDDI to the data center. |
| 03:00.02 | slav3_kitten | nice |
| 03:00.19 | fubada | im forcing my iface to 100full even though it negotiates at 100half |
| 03:00.29 | fubada | theres some interface errors with 100half |
| 03:00.40 | jpsharp | Force both interfaces if you can. |
| 03:00.51 | fubada | cant control the managed router |
| 03:00.55 | fubada | forcing only my * side |
| 03:01.09 | jpsharp | Hopefully the other side autos to 100/Full. |
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| 03:01.33 | fubada | it doesnt |
| 03:01.40 | fubada | oh |
| 03:01.52 | fubada | right..wouldnt it just not work ? |
| 03:01.58 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:02.04 | slav3_kitten | fubada, check your cables |
| 03:02.09 | slav3_kitten | something isn't right |
| 03:02.52 | slav3_kitten | two wrongs don't make a right |
| 03:02.52 | fubada | the t1 router insist on 100half when i auto negotiate |
| 03:03.12 | fubada | i can set my * side to 100full and it establishes the link at 100full |
| 03:03.27 | fubada | and 100half i get some interface errors, roughly 400 errors in 5min time |
| 03:03.29 | fubada | on TX |
| 03:03.33 | slav3_kitten | this sounds like bad wiring. |
| 03:03.38 | fubada | at 100full - none |
| 03:03.48 | jpsharp | No. Sometimes autonegotation doesn't. |
| 03:03.56 | fubada | w checked cables |
| 03:03.57 | jpsharp | I've seen it time & time again. |
| 03:04.09 | slav3_kitten | jpsharp, i've had that exactly 3 times in my life, bad punchdowns were the result |
| 03:04.21 | fubada | this is direct cable |
| 03:04.30 | jpsharp | Ive seen it on direct cables. |
| 03:04.52 | jpsharp | No matter what cables I used, the interfaces wouldn't agree. |
| 03:05.04 | fubada | they agree on 100half |
| 03:05.29 | slav3_kitten | shrugs |
| 03:05.30 | fubada | with ethtool on * side i can set 100full |
| 03:05.37 | slav3_kitten | i've far from seen everything |
| 03:05.46 | fubada | thanks guys |
| 03:07.51 | jpsharp | Ive been doing this for damn near 20 years |
| 03:08.17 | slav3_kitten | i'm sorry |
| 03:08.33 | jpsharp | i enjoy it |
| 03:09.12 | slav3_kitten | if you love what you do, never work a day right? |
| 03:09.20 | jpsharp | Something like that. |
| 03:09.42 | slav3_kitten | i love it as a hobby not as a career |
| 03:09.50 | slav3_kitten | but i'm looking into it as a career now |
| 03:10.02 | jpsharp | I've been doing it for almost 20 years and I'll be 38 in a few months. |
| 03:10.20 | last1 | what can cause garbled audio after pressing a dtmf tone ( key being pressed ) ? |
| 03:10.35 | jpsharp | Inband DTMF? What codec? |
| 03:10.52 | last1 | it's inband dtmf, ulaw |
| 03:11.00 | slav3_kitten | you got 10 years on me roughly |
| 03:11.06 | last1 | so I have a file playing in the background |
| 03:11.29 | last1 | and if i press 1 for example, the file stops playing ( which is the right behavior ) but then I hear a little static/crackling at the end |
| 03:11.37 | last1 | almost as if it didn't stop playing properly |
| 03:12.21 | jpsharp | Does the crackling continue or is it just a short time after the file stops? |
| 03:12.27 | last1 | just a very short time |
| 03:12.29 | last1 | like 1/2 a second |
| 03:12.32 | last1 | but it's very annoying |
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| 03:13.40 | last1 | so if the audio says: "Press 1 for tech support" and you press 1 right after "Press" you will hear: Press<beep><crackle> |
| 03:13.56 | last1 | the beep being the dtmf tone for 1 |
| 03:15.08 | last1 | what makes that crackling noise and how can I remove it ? |
| 03:17.30 | jpsharp | What format are your voice files in? |
| 03:17.54 | jpsharp | Maybe a hiccup from changing codecs after playback? |
| 03:29.11 | last1 | the files are in wav format |
| 03:29.21 | last1 | they sound fine on my pc |
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| 04:24.57 | PhoenixMage | Anyone able to remind me of the key combo to reboot a cisco phone? |
| 04:26.52 | PhoenixMage | Found it |
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| 05:24.07 | Dovid | is there any way to do a catch all extension with _ ? |
| 05:24.08 | Dovid | meaning |
| 05:24.23 | Dovid | Exten => conference[a-zA-Z].,1,Noop |
| 05:24.36 | Dovid | and the extension calld ies conference_all_users? |
| 05:24.40 | Dovid | that does not seem to work |
| 05:25.37 | jpsharp | exten => _.,1,Foo catches everything. |
| 05:25.45 | jpsharp | Usually a lot of stuff you don't want. |
| 05:28.43 | kaldemar | Dovid: [a-zA-Z] means 1 character of a-z or A-Z |
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| 05:39.37 | Dovid | kaldemar: I know that. I wanted to know how to incluude _ as well |
| 05:39.53 | Dovid | jpsharp: Asterisk does not play nicely when you use _., |
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| 05:40.05 | Dovid | also i only want extensions prefixed with conference_ |
| 05:40.11 | Dovid | to got to a certain context |
| 05:44.35 | kaldemar | Dovid: than add _ inside your [] |
| 05:56.04 | Dovid | kaldemar: Tried it didnt work |
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| 07:02.12 | *** join/#asterisk Yourname` (~whocares@unaffiliated/yourname/x-837320) |
| 07:04.43 | Yourname` | Hello. I connect to our ITSP over their IP-VPN link. Internally in the office, we're 192.168.1.x.. but for the ITSP network, we're 192.168.200.x. Problem is now we have remote agents, so we had to set sipnat settings like externip, etc. However, for calls going through the ITSP, we still need 192.168.200.x in the sip packets. Problem is, due to externip (to avoid audio issues between remote agents/nat pbx) the ip being sent to ITSP i |
| 07:04.44 | Yourname` | s our externip. How can I make it so that ITSP receives our 192.168.200.2 IP? |
| 07:19.14 | kaldemar | Yourname`: does your asterisk box have a 192.168.200 address? |
| 07:19.35 | Yourname` | kaldemar: Yes. It has 192.168.1.252 for local lan. And 192.168.200.2 for the ITSP\ |
| 07:22.17 | kaldemar | define the 192.168.200 network as localnet |
| 07:23.51 | Yourname` | kaldemar: I did.. in the localnet portion.. added 192.168.1.0 as well.. |
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| 07:44.28 | kaldemar | Yourname`: how? show your configs and a sip debug of a call. |
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| 08:18.46 | v0lZy | lo |
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| 08:42.20 | bulkorok | hi |
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| 09:40.00 | CommaCrazy | could some 1 elaborate on this sentence for VoIPex "Improves user experience increasing rate of adoption" |
| 09:44.16 | kaldemar | CommaCrazy: blah blah blah, yadda yadda yadda. |
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| 09:53.01 | CommaCrazy | kaldemar, so basically nothing :P |
| 10:06.03 | bulkorok | is it possible to write FAXOPT(error) in a database-cloumn via cdr_adaptive_odbc? |
| 10:10.59 | *** join/#asterisk [SySteM] (~antoine@195.214.230.1) |
| 10:11.05 | [SySteM] | Hello, i work for a radio station |
| 10:11.11 | [SySteM] | is it possible to use asterisk |
| 10:11.17 | [SySteM] | with very good codec |
| 10:11.23 | [SySteM] | for have the better voice quality ? |
| 10:11.53 | [SySteM] | why not music ? i search the less latency possible |
| 10:12.08 | kaldemar | CommaCrazy: "our product is good, people will want to use it" |
| 10:12.22 | kaldemar | CommaCrazy: so nothing much technically. |
| 10:17.49 | CommaCrazy | kaldemar, that's what I taught but had to check :D |
| 10:43.31 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 10:43.35 | [sr] | iellow! |
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| 10:48.06 | *** join/#asterisk din3sh (~din3sh@41.212.200.24) |
| 10:48.14 | din3sh | hello all |
| 10:54.58 | *** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za) |
| 10:55.50 | like_a_horse | hi all, how would i go about unsetting a variable. Set(ARG2="") seems a little clunky |
| 10:56.57 | bulkorok | do you want to delete tha var or just empty it!? |
| 10:58.44 | like_a_horse | delete would be better i think |
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| 11:03.35 | bulkorok | in the deeps I found: Setting it to an empty string effectively resets the value. |
| 11:04.58 | like_a_horse | ok |
| 11:05.11 | like_a_horse | so ARG2="" is the best i could hope for? |
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| 11:05.37 | bulkorok | seems like... but if its a channel variable it will be destroyed after hangup anyway... |
| 11:06.13 | like_a_horse | ok cool |
| 11:06.15 | like_a_horse | thanks :) |
| 11:06.25 | bulkorok | :) |
| 11:14.52 | like_a_horse | bulkorok, check out http://pastebin.com/dPj8tbvT |
| 11:15.26 | like_a_horse | looks like i should go for Set(ARG2=) instead |
| 11:15.27 | like_a_horse | :) |
| 11:16.08 | bulkorok | perfect! |
| 11:16.20 | kaldemar | like_a_horse: if you're using a version newer than 1.4.X and don't have "app_set=1.4" in asterisk.conf |
| 11:17.47 | like_a_horse | kaldemar, oh, does Set(ARG2=) not work in 1.4 ? |
| 11:18.12 | like_a_horse | running 1.6.2 here |
| 11:18.15 | kaldemar | like_a_horse: i didn't say so. :) |
| 11:18.15 | like_a_horse | :) |
| 11:18.48 | kaldemar | like_a_horse: quotes are interpreted as characters in post-1.4 app Set. |
| 11:18.58 | like_a_horse | aha |
| 11:18.59 | like_a_horse | i see |
| 11:19.03 | like_a_horse | good to know |
| 11:19.33 | kaldemar | you don't need the quotes in earlier versions either. |
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| 11:24.17 | like_a_horse | cool |
| 11:24.18 | like_a_horse | thanks |
| 11:24.19 | like_a_horse | :) |
| 11:36.12 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 11:39.13 | aberrios | Qwell: You here? I have a question about the network gateway monitoring in pacemaker for the R-Series devices.. So this section (lines 13-15) in the sample pacemaker.cfg (http://bit.ly/LbuO7f) monitors the network gateway (usually a router)... If a user were to restart their router for whatever reason, would this cause a migration of services? |
| 11:40.19 | aberrios | Reason I ask is, I've had three sites where the cluster started to migrated to a fro and neither node would fully start up its stack. At each site they had restarted their routers (network gateways). |
| 11:40.46 | aberrios | I had to restart both nodes fully after the gateway was up and it settled down. |
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| 11:57.43 | rgsteele | For some reason, whenever my dialplan tries to play moh sounds, I get this repeatedly in the logs: res_musiconhold.c: poll() failed: Interrupted system call |
| 11:58.16 | rgsteele | I can't find any other debugging info that would lend a clue as to exactly what's causing the problem. |
| 11:59.01 | rgsteele | Is there something else I can try, or somewhere I can check for more info? Teh Googlez isn't being very helpful. |
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| 12:01.35 | rgsteele | (On a side note - is something going on w/ issues.asterisk.org? I'm having trouble loading the site.) |
| 12:07.52 | kaldemar | rgsteele: try issues.asterisk.org/jira/ |
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| 12:11.10 | rgsteele | Seems to work, thanks kaldemar |
| 12:12.28 | twanny796 | reading the third edition of the * book, cannot find how to make an IAX connection?? anyone knows which page? |
| 12:13.52 | rgsteele | sighs |
| 12:14.01 | rgsteele | CDR hell is worse than normal hell :-/ |
| 12:16.17 | kaldemar | twanny796: the 3rd edition doesn't really cover much of IAX. |
| 12:16.47 | rgsteele | Hmm.... I wonder if the use of Goto breaks billsec and disposition in 1.8 as well as 1.6.... (https://issues.asterisk.org/jira/browse/ASTERISK-16330) |
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| 12:25.25 | twanny796 | kaldemar: yes, referring to the second ;) |
| 12:26.43 | rgsteele | Anybody know of a good reference for CDR debugging? |
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| 12:36.08 | PhoenixMage | hi guys, not sure if this is the best place to ask but I am playing arounf with .cnf.xml files on a cisco phone. It is using skinny but do I still configure things like the buttons under <sipProfile>? |
| 12:40.02 | rgsteele | Oh interesting... for the inbound call, once the queue agent answers the call that's in the queue, the callerID on the agent's channel is set to his extension... the caller ID from the incoming call only exists on the channel that's open to the SIP provider. Even though the agent's channel is BridgedTo the SIP provider's channel. |
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| 12:41.50 | rgsteele | The application is set to AppQueue instead of BridgedCall on the agent's channel according to a 'core show channels verbose'... so maybe it's not just CDR's fault. |
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| 12:43.47 | rgsteele | Oh, and instead of the extension being set to nothing in the version of Asterisk where CDR shows billsec/disposition correctly, it's set to the extension that the inbound caller hit to get into the desired queue (sales/support/etc.). I guess that's where it loses it - once that extension is dialed. |
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| 13:13.18 | *** join/#asterisk pithagorians (~pithagori@217.12.122.18) |
| 13:13.51 | pithagorians | hi. did anybody used md5secret in asterisk? |
| 13:16.02 | pithagorians | when i use md5secret=321d447b895ec9613714c6dfb597d37b as generated like echo -n "ih7fi<$JQI" | md5sum and use the ih7fi<$JQI password in phone settings registration string comes with 'Wrong password' |
| 13:19.26 | mamikk | pithagorians, thats not the correct formula for md5secret, it is md5(username:real:password) -- http://en.wikipedia.org/wiki/Digest_access_authentication |
| 13:19.52 | mamikk | realm* that is |
| 13:20.40 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 13:21.00 | pithagorians | solved |
| 13:21.03 | pithagorians | thx |
| 13:21.14 | pithagorians | the right format is echo -n "<user>:<realm>:<secret>" | md5sum |
| 13:21.22 | pithagorians | i used only the pass |
| 13:22.38 | slav3_kitten | interesting |
| 13:23.30 | *** join/#asterisk tfgtech (~tfgtech@219.Red-83-42-95.dynamicIP.rima-tde.net) |
| 13:24.41 | pithagorians | i want to use real time functionality of asterisk |
| 13:24.48 | pithagorians | and i have a database of users |
| 13:25.08 | pithagorians | with passwords in md5 formats |
| 13:25.12 | slav3_kitten | so because i'm a total noob and learning asterisk... |
| 13:25.13 | pithagorians | format* |
| 13:25.25 | slav3_kitten | what's the end goal you're trying to accomplish? |
| 13:25.28 | pithagorians | because of that i need it |
| 13:25.37 | pithagorians | never mind |
| 13:25.39 | pithagorians | :) |
| 13:25.44 | pithagorians | i just can tell you |
| 13:25.50 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jihxbkmxztaptqhi) |
| 13:25.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:26.04 | pithagorians | if it's interesting for you |
| 13:26.07 | slav3_kitten | it is |
| 13:26.14 | pithagorians | ok |
| 13:26.20 | slav3_kitten | i like learning stuff, and it seems like you're doing something pretty interesting |
| 13:26.43 | pithagorians | so as i said asterisk can keep it's sip accounts in a data base |
| 13:26.52 | [TK]D-Fender | MD5 password doesn't really qualify as "intersting"... nor does jsut having SIP peers in a DB.... |
| 13:27.04 | pithagorians | like mysql, postgresql |
| 13:27.05 | slav3_kitten | [TK]D-Fender, shush i'm learning here :P |
| 13:27.27 | [TK]D-Fender | slav3_kitten, Then you've just lerned that 2 more things aren't "interesting" ;) |
| 13:27.37 | pithagorians | :D |
| 13:27.38 | slav3_kitten | interesting to me |
| 13:27.57 | slav3_kitten | so anyhow, user database |
| 13:28.07 | pithagorians | yes |
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| 13:30.52 | pithagorians | for you just remember that you can manage your users easy when you have a lot and keep their passwords in md5 hashes instead of plain text |
| 13:36.43 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 13:39.01 | slav3_kitten | so basically it's just real time user management? |
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| 13:42.43 | *** join/#asterisk scgm11 (~Sebastian@187-14-75-79.user.veloxzone.com.br) |
| 13:43.56 | scgm11 | hi I'm having a problem compiling dahdi 2.5.0.2 on ubuntu 12.04 server |
| 13:44.41 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 13:44.42 | scgm11 | http://pastebin.com/mMVhLRpK |
| 13:44.48 | scgm11 | anyone can help? |
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| 13:50.04 | slav3_kitten | one sec |
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| 13:52.15 | *** join/#asterisk os_Florent (~chatzilla@62.244.88.2) |
| 13:52.21 | slav3_kitten | https://bugs.launchpad.net/ubuntu/+source/dahdi-linux/+bug/928499 |
| 13:52.41 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 13:53.31 | slav3_kitten | scgm11, any reason you're building 2.5.0.2? |
| 13:53.42 | scgm11 | ok i will have to patch manually for this due to the version I'm using is of a card provider in brazil |
| 13:53.45 | slav3_kitten | latest is 2.6.1+2.6.1 |
| 13:53.55 | os_Florent | Can somebody tell me if it possible to know from where an hangup request come from in asterisk dialplan/AGI |
| 13:54.18 | scgm11 | I know works ok with 2.6 |
| 13:54.23 | scgm11 | but i need this version now |
| 13:54.59 | slav3_kitten | well i can't really help you, sorry |
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| 13:56.01 | scgm11 | now you help a lot now I just have to patch this version with this diff |
| 13:56.08 | scgm11 | and i think should work |
| 13:57.08 | *** join/#asterisk timahvo1 (~rogue@41.215.82.210) |
| 13:57.35 | d_preston215 | When creating a callfile, what channel do I give it if I want it to go to a conference? |
| 13:57.46 | hwt | hi, when I start asterisk from init script and run an AGI script that should print console messages through STDERR, it doesn't work. any ideas? |
| 13:57.54 | hwt | it works when I start asterisk with asterisk -cvv |
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| 14:08.26 | os_Florent | Hi , it is possible to know which channel hangup a call in diaplan ? |
| 14:15.10 | vlad_starkov | Hello there! What is supported subset of WAV bitrates for Asterisk 10? |
| 14:16.42 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:22.47 | scgm11 | slav3_kitten: that did the trick!!! Im able to compile! |
| 14:23.02 | slav3_kitten | i am just that good? |
| 14:23.20 | scgm11 | yes! and I was able to use that patch jeje |
| 14:23.48 | slav3_kitten | so why do you have to use the older version? |
| 14:24.08 | scgm11 | because that version is downloaded from a card manufacturer in brazil |
| 14:24.12 | scgm11 | aligera cards |
| 14:24.19 | scgm11 | and they are using that version |
| 14:24.24 | scgm11 | and is patched for their cards |
| 14:24.42 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:24.54 | scgm11 | I think i could also patch new version with their patch |
| 14:25.03 | scgm11 | but that could take much time |
| 14:25.10 | scgm11 | I needed this now! |
| 14:25.14 | WIMPy | Cards with driver issues really suck. |
| 14:25.20 | scgm11 | yes! |
| 14:25.40 | slav3_kitten | ah got ya |
| 14:25.51 | WIMPy | Didn't we all learn that you should steer clear of hardware that isn't supported by Linux natively already? |
| 14:25.51 | scgm11 | or maybe just one .c |
| 14:25.53 | scgm11 | to include |
| 14:26.08 | scgm11 | this client already has that card |
| 14:26.19 | scgm11 | I always use digium cards :) |
| 14:26.28 | slav3_kitten | WIMPy, i thought we learned buying hardware unsupported leads to support being developed |
| 14:26.49 | WIMPy | I didn't. |
| 14:27.13 | slav3_kitten | lol |
| 14:28.05 | [TK]D-Fender | os_Florent, Dial() option : g |
| 14:28.21 | os_Florent | i just saw this |
| 14:28.29 | os_Florent | thx |
| 14:28.40 | slav3_kitten | ponders calling this cisco hardware provider an seeing if he can socially engineer himself a 7921... on the super cheap |
| 14:29.23 | slav3_kitten | [TK]D-Fender, know much about the skinny support in asterisk? |
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| 14:36.12 | [TK]D-Fender | slav3_kitten, Virtually nothing, other than you shouldn't avoid. |
| 14:36.16 | [TK]D-Fender | should* |
| 14:36.34 | [TK]D-Fender | Stick with real commodity protocols |
| 14:36.38 | slav3_kitten | well that's not a help :/ |
| 14:37.09 | slav3_kitten | i already have cisco gear, an i really like their message waiting indicator light pipe in the handset |
| 14:37.14 | [TK]D-Fender | We're helping you move towards better solutions |
| 14:37.17 | slav3_kitten | mainly i already have it |
| 14:37.22 | slav3_kitten | and i'm piss poor |
| 14:37.38 | slav3_kitten | i kinda really want a polycom phone |
| 14:37.40 | [TK]D-Fender | slav3_kitten, so reflash them to SIP <--------- |
| 14:37.40 | WIMPy | has the impression that SCCP is a lot better than SIP. |
| 14:37.50 | WIMPy | But then, I guess, anything is. |
| 14:38.05 | slav3_kitten | [TK]D-Fender, can't reflash the 7921g |
| 14:38.19 | slav3_kitten | it does sccp or sccp or sometimes sccp |
| 14:38.53 | slav3_kitten | the 7911, 7940, an 7960 have sip firmware but my cisco buddy told me they suck balls |
| 14:39.06 | [TK]D-Fender | slav3_kitten, So then use them.... |
| 14:39.16 | WIMPy | Is that a sales argument? |
| 14:39.26 | slav3_kitten | WIMPy, that was his sales argument |
| 14:39.40 | slav3_kitten | mostly i'm concerned about the 7921 that i don't have in my posession yet |
| 14:39.50 | [TK]D-Fender | Asking the Cisco guy about NON-Cisco is like asking a Burger King cashier if Big Mac's are better.... |
| 14:39.52 | _Corey_ | slav3_kitten: The SIP fimware on those models works fine, it's just not developed much in the last 5 or so years |
| 14:40.29 | slav3_kitten | _Corey_, they are phones, how much can you really improve on them |
| 14:40.41 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 14:40.51 | WIMPy | Quite a lot, I'm sure. |
| 14:40.52 | slav3_kitten | but i plan on playing with both sip an sccp on them if i can when i get time |
| 14:41.03 | _Corey_ | slav3_kitten: Well, most of the firmware has evolved a bit to include new features, streamline provisioning, etc etc... that stuff not so much |
| 14:41.17 | slav3_kitten | i wonder what hotel house cleaning is going to think of this room when they come in today |
| 14:42.20 | *** join/#asterisk lsilva_adclick (~lsilva@188.250.13.250) |
| 14:42.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:42.50 | Nugget | Using cisco phones with asterisk is not recommended. |
| 14:43.07 | Nugget | they're gorgeous but it's like wearing a hair shirt |
| 14:43.17 | os_Florent | [TK]D-Fender, dial() option : g works, but if i use AGI there id any kind of variable like HANGUPDIRECTION |
| 14:43.45 | [TK]D-Fender | $(DIALSTATUS} <----------- |
| 14:43.47 | slav3_kitten | Nugget, plus side is it's not a production setup, just a home dicking about to learn asterisk thing |
| 14:44.09 | Nugget | slav3_kitten: bear in mind that the 20/40/60 phones use a completely different configuration format than the newer phones |
| 14:44.11 | WIMPy | Do the real Cisco phones have an own SIP implemenataion or are they also using recycled Sipura stuff? |
| 14:44.21 | [TK]D-Fender | Their own |
| 14:44.22 | Nugget | WIMPy: entirely their own |
| 14:44.29 | *** join/#asterisk GameGamer43 (~users.553@gateway/web/irccloud.com/x-wjeetbukcjlxiljz) |
| 14:44.32 | *** join/#asterisk NotHere (~steve@geozix.static.otenet.gr) |
| 14:44.32 | [TK]D-Fender | SPA = Sipura |
| 14:44.46 | Nugget | slav3_kitten: so much of your googling will turn up information that's not useful. if you see "xml" it's not relevant to your phones |
| 14:44.50 | WIMPy | Yes, the SPAs suck. |
| 14:44.54 | [TK]D-Fender | Which... when you think of it... is pretty much Cisco in it's own way. |
| 14:45.18 | slav3_kitten | Nugget, even with the latest firmware on the 20/40/60 phones? |
| 14:45.26 | Nugget | correct |
| 14:45.34 | slav3_kitten | i was reading someplace that the newest firmware uses xml ... |
| 14:45.39 | NotHere | since upgrading to asterisk 1.8.13 from 1.8.10, asterisk segfaults and restarts in many occasions |
| 14:45.44 | Nugget | for the 41/61/45/65 phones etc |
| 14:45.49 | lsilva_adclick | Hi, I am using a voip system with asterisk already installed with a svn version r1826M. Can anyone tell me how to figure out the date of that revision? |
| 14:46.24 | slav3_kitten | still... i have the hardware an no means to buy new polycom or other vendor phones |
| 14:46.40 | WIMPy | lsilva_adclick: Err, 4-digit version? That mus be from last century. |
| 14:46.51 | NotHere | when I try to transfer a call to another extension, when the other extensions rings, asterisk restarts (and leaves a mess, where the call is lost...) |
| 14:47.07 | lsilva_adclick | yep, those are my suspicious |
| 14:47.37 | lsilva_adclick | i'm having so many problems and can't update the box, was afraid that was the case |
| 14:48.24 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 14:48.45 | WIMPy | lsilva_adclick: I'd imagine such an old version not having many issues, because it can't do much, anyway. |
| 14:49.39 | slav3_kitten | Nugget, can i bother you a bit in pm about cisco phones? |
| 14:49.48 | Nugget | sure |
| 14:51.50 | lsilva_adclick | well, my main issue are random call disconnects not related with extension, time of day, trunk or any other factor that I can figure out |
| 14:52.38 | lsilva_adclick | and trying to get internal_timing on makes the asterisk process reboot |
| 14:52.46 | WIMPy | You should get another box and do a new install. This is going to be a big change. |
| 14:53.06 | WIMPy | It's from 2003-12-03, BTW. |
| 14:53.13 | [TK]D-Fender | lsilva_adclick, there is no trace for a drop like that. |
| 14:53.28 | lsilva_adclick | thanks for the date, how did you get there? |
| 14:53.32 | [TK]D-Fender | lsilva_adclick, If it dies due to a failure it'll look like the remote side ended it. |
| 14:53.50 | WIMPy | lsilva_adclick: Searched the svn log. |
| 14:54.03 | lsilva_adclick | that's what I thought until i realised it happens about four or five times a day |
| 14:54.05 | [TK]D-Fender | 2003? Holy crap |
| 14:54.16 | [TK]D-Fender | that's insane.. |
| 14:54.35 | WIMPy | That must be the oldest version, we have heard about in here. |
| 14:54.53 | lsilva_adclick | i decided to buy a yeastar mypbx pro because i'm a newbie with voip |
| 14:55.23 | [TK]D-Fender | lsilva_adclick, ok/fine/sure |
| 14:55.26 | lsilva_adclick | just thought it would be easier and saffer to go with something already built with newbies in mind |
| 14:55.43 | lsilva_adclick | but it turns out it's the other way around |
| 14:56.14 | lsilva_adclick | just a pain in the *** to change a simple option in asterisk.conf |
| 14:56.41 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:57.05 | [TK]D-Fender | What do you even have to change in asterisk.conf? |
| 14:57.18 | [TK]D-Fender | None of my installs ever needed to touch it |
| 14:57.25 | lsilva_adclick | enabling the internal_timing |
| 14:57.26 | [TK]D-Fender | (from the sample provided) |
| 14:57.33 | [TK]D-Fender | that's a pain in the ass? |
| 14:57.58 | [TK]D-Fender | 2003 didn't even have that option.... |
| 14:58.07 | lsilva_adclick | lool hell yeah, it gets rewritten whenever I change something in the frontend |
| 14:58.09 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 14:58.16 | lsilva_adclick | yeastar thing |
| 14:58.19 | leifmadsen | that's how web interfaces typically work |
| 14:58.31 | leifmadsen | if you're going to use a web interface, you're not supposed to touch the files manually |
| 14:58.43 | leifmadsen | if you need the web to do something different, you need to modify the webapp |
| 14:58.51 | lsilva_adclick | a core restart now does the same, a rewrite of asterisk.conf |
| 14:58.58 | leifmadsen | nope |
| 14:59.09 | [TK]D-Fender | * does not write that file |
| 14:59.13 | leifmadsen | restarting asterisk itself does not rewrite any configuration files |
| 14:59.16 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 14:59.27 | WIMPy | Err, did I understand that right. The thing you bought was a 9 year old Asterisk Version with a frontend? |
| 14:59.37 | lsilva_adclick | basically |
| 14:59.52 | WIMPy | hardcore |
| 14:59.53 | lsilva_adclick | i should return that crappy thing |
| 15:00.13 | [TK]D-Fender | Please place your head flush with the barrel and go stand over there on that large plastic sheet. Thanks ;) |
| 15:00.16 | WIMPy | hardcore ripoff, that is |
| 15:01.26 | [TK]D-Fender | And yes, you should return it for as full a refund as you can get. |
| 15:01.36 | lsilva_adclick | I don't really need anything fancy but something that worked fine with be nice |
| 15:01.39 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1828.pool.mediaWays.net) |
| 15:02.29 | lsilva_adclick | Would you consider trixbox or installing some linux and asterisk on top of it? |
| 15:02.34 | [TK]D-Fender | NO |
| 15:02.40 | [TK]D-Fender | Trixbox was dead 2 YEARS ago |
| 15:03.11 | lsilva_adclick | ok, that's how uneducated I am about this |
| 15:03.24 | WIMPy | You just learned about what GUIs do yo you. But if you really want one, choose one that's still maintained. |
| 15:03.30 | [TK]D-Fender | The only 2 more ready-to-go distro's really worth it right now are AsteriskNOW, and FreePBX's ISO. |
| 15:03.37 | [TK]D-Fender | Both Offer FreePBX as the management GUI |
| 15:03.57 | leifmadsen | elastix if you want a package seems to be maintained |
| 15:04.00 | lsilva_adclick | And can I get FXO cards to work well with those? |
| 15:04.02 | leifmadsen | trixbox is... old news |
| 15:04.13 | WIMPy | Gemeinschaft also offer an ISO. |
| 15:04.18 | [TK]D-Fender | Elastix rus a forked FreePBX and will cause issues as you go though... |
| 15:04.23 | [TK]D-Fender | runs* |
| 15:04.32 | Nugget | "99 distros of asterisk on the wall, 99 distros of asterisk... take one down, pass it around... 112 distros of asterisk on the wall..." |
| 15:04.39 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:04.51 | rgsteele | When an incoming call is bridged to an agent in a queue, should the agent's channel inherit the CDR variables set in the inbound channel? Or, if it doesn't by default, is there some config option that will help achieve that? For some reason, when the call is bridged, the agent's channel's CDR variable "disposition" indicates the call wasn't answered, even though it was, and I'm wondering... |
| 15:04.53 | rgsteele | ...if something is locking the CDR & preventing it from updating. |
| 15:05.02 | [TK]D-Fender | Nugget, Only 3 kinds of people in this world.. those that know math, and those that don't. |
| 15:05.50 | Nugget | [TK]D-Fender: if you can understand hexidecimal, you're 1 in a |
| 15:06.13 | *** join/#asterisk alexscott (~alexscott@37.122.202.194) |
| 15:06.29 | [TK]D-Fender | Nugget, only 10 kinds of people in the world... those that understand binary, and those that don't |
| 15:06.37 | os_Florent | [TK]D-Fender, dial() option : g works, if u use dialplan. Is there any way to know this before existing an AGI which lauch the call |
| 15:07.01 | [TK]D-Fender | look at ${HANGUPCAUSE} |
| 15:08.08 | WIMPy | would expect 16 in both cases. |
| 15:09.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:09.54 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 15:10.05 | lsilva_adclick | From what I've been reading, running asterisk on VM isn't very good idea either right? something to do with timing issues |
| 15:10.44 | _Corey_ | lsilva_adclick: That's outdated information |
| 15:11.02 | rgsteele | I have an inbound call, it goes into a queue, and then bridges to an agent in the queue. But heavy debugging output shows this in the logs: "[Jun 7 10:50:00] DEBUG[30104] chan_sip.c: Outgoing Call for agentname". Why in the world would it label the inbound call as an outgoing call? |
| 15:12.11 | lsilva_adclick | I have a server I can use running vmware hypervisor. I could try to run from there, but just didn't want to be another fluke |
| 15:12.39 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:12.50 | os_Florent | [TK]D-Fender, Yes ${HANGUPCAUSE} work but i wana know from which side the call has been hangup not only the cause, For example in a succesfull call ${DIALSTATUS==answer and ${HANGUPCAUSE}=16 but i dont know who hangUP |
| 15:14.55 | JamKo | Is it really necessary to stagger the sip ports on a multi-line phone (5060,5061 etc)? As long as the firewall is able to successfully nat the phone's IP, shouldn't the phone be able to sort out which line gets the data? |
| 15:15.03 | _Corey_ | lsilva_adclick: I have some largish systems running on ESXi (500+ users) |
| 15:15.04 | Qwell | JamKo: no... |
| 15:15.22 | Qwell | any guide that says to use anything other than 5060 needs to be launched into orbit |
| 15:15.33 | *** join/#asterisk din3sh (din3sh@41.136.86.62) |
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| 15:15.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:16.52 | JamKo | Qwell: thanks lol. The cisco and old linksys spa phones, stagger the ports by default, and I always found that to be an odd thing to do. |
| 15:17.16 | Qwell | You had me at Cisco. |
| 15:18.01 | slav3_kitten | man it's like talk about cisco day :D |
| 15:18.07 | slav3_kitten | look what i started |
| 15:19.57 | lsilva_adclick | <PROTECTED> |
| 15:20.37 | rgsteele | Meh. I feel like telling the phone reps to write down call answer/hangup times on a piece of paper. Wasting days on this CDR stuff is teh suck :-/ |
| 15:20.42 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
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| 15:21.04 | lsilva_adclick | do you have an digium card on it? |
| 15:21.24 | _Corey_ | lsilva_adclick: No, when virtualized you need to connect your hardware elsewhere |
| 15:21.31 | NotHere | when I try to transfer a call to another extension, when the other extension rings, asterisk restarts (/usr/sbin/safe_asterisk: line 145: 8604 Segmentation fault (core dumped) and exit status 139, signal 11) |
| 15:22.19 | NotHere | that happens from asterisk 1.8.11.1 to 1.8.13 and last version which works correctly is 1.8.10.1 |
| 15:23.20 | WIMPy | What channeltypes? |
| 15:23.21 | slav3_kitten | wonders... |
| 15:23.29 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 15:23.36 | lsilva_adclick | thanks for the info |
| 15:24.08 | *** part/#asterisk lsilva_adclick (~lsilva@188.250.13.250) |
| 15:24.35 | NotHere | WIMPy, sip phones |
| 15:25.29 | [TK]D-Fender | os_Florent, If yuo are in the next priority... and the result was ANSWER... then you should read what "g" does again |
| 15:30.00 | aberrios | Qwell: did you see my message a few hours ago, re: pacemaker.cfg ? |
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| 15:39.06 | Qwell | aberrios: no |
| 15:39.19 | aberrios | Qwell: ill just paste it again...one mo |
| 15:39.23 | aberrios | I have a question about the network gateway monitoring in pacemaker for the R-Series devices.. So this section (lines 13-15) in the sample pacemaker.cfg (http://bit.ly/LbuO7f) monitors the network gateway (usually a router)... If a user were to restart their router for whatever reason, would this cause a migration of services? |
| 15:39.28 | aberrios | Reason I ask is, I've had three sites where the cluster started to migrated to a fro and neither node would fully start up its stack. At each site they had restarted their routers (network gateways). |
| 15:39.32 | aberrios | I had to restart both nodes fully after the gateway was up and it settled down. |
| 15:40.12 | Qwell | Both boxes should have seen the same timeout, and nothing would have changed. |
| 15:40.28 | Qwell | but, sure, it's possible |
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| 15:42.02 | aberrios | Qwell: Would removing this section from pacemaker.cfg be advisable? I'm going to test this on a the bench soon (in the middle of changing jobs and offices....). |
| 15:42.35 | Qwell | sure, if you don't care whether they can access things |
| 15:42.47 | Qwell | it also doesn't *have* to be your gateway address |
| 15:43.01 | Qwell | it's just there to make sure that connectivity is still there |
| 15:43.59 | aberrios | Qwell: righto. Its just the cluster started having a major wobbler when the sites restarted their routers...Definately one to test on a bench. |
| 15:44.21 | Qwell | Don't do that then. :) |
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| 15:46.15 | aberrios | Would you like me to post my findings anywhere? I guess a lot of sites would restart their gateways at some point, and having a HA cluster that wobbles when the gateway restarts isn't ideal.... |
| 15:46.53 | Qwell | Having an HA cluster where a gateway restarts isn't ideal. That should be considered a problem. |
| 15:47.17 | aberrios | So you're saying Gateway and Router should be seperated in an ideal installation? |
| 15:47.32 | Qwell | I'm saying that for HA, those things don't go down - ever. |
| 15:48.09 | Qwell | If you want something less than that, you're able to use a different address. Or you can increase the timeouts. Loads of stuff you could do |
| 15:48.46 | aberrios | I could use the IP of a managed switch they're connected to |
| 15:49.02 | Qwell | sure |
| 15:49.11 | Qwell | as long as it's something they can both reach |
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| 16:30.51 | *** join/#asterisk nsgn (~nsgn@cpe-24-28-31-68.austin.res.rr.com) |
| 16:31.37 | nsgn | help! thanks to some fumble fingered users i've got two bad dynamic agents stuck in a queue. i cant seem to get them out with queue remove member. the agents show up in queue show as "*@from-queue" and "109*@from-queue" |
| 16:31.44 | nsgn | and with them in there the entire queue somehow fails to take calls |
| 16:32.01 | nsgn | rebooting the asterisk box does not remove them from the queue as all dynamic members seem to be remembered |
| 16:32.24 | [TK]D-Fender | So go remove them |
| 16:32.43 | [TK]D-Fender | RemoveQueueMember <- |
| 16:32.49 | nsgn | when i do queue remove member blah blah it just comes back and says "not in queue" |
| 16:33.20 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 16:33.27 | nsgn | but i've never had to do this before so perhaps my usage is wrong? |
| 16:33.28 | [TK]D-Fender | Show us |
| 16:33.29 | [TK]D-Fender | ~pb |
| 16:33.29 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:33.30 | [TK]D-Fender | ^^^ |
| 16:36.40 | nsgn | [TK]D-Fender, http://pastebin.com/8BPzYqPq |
| 16:37.29 | [TK]D-Fender | nsgn, that is not an INTERFACE |
| 16:37.32 | [TK]D-Fender | there is no TECH in there |
| 16:37.40 | [TK]D-Fender | and you should have included the queue dump with that |
| 16:38.30 | nsgn | sorry, here is queue show |
| 16:38.31 | nsgn | http://pastebin.com/iT6QhijF |
| 16:39.23 | nsgn | see the two bad ones in there? |
| 16:39.41 | [TK]D-Fender | Local/109*@from-queue/n <------- member |
| 16:41.16 | nsgn | ah. i'm stupid. one moment while i test functionality is restored |
| 16:44.06 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 16:44.17 | MichaelB8225 | Good morning |
| 16:47.39 | nsgn | [TK]D-Fender, thanks. functionality restored, users scolded, config tweaks made to avoid in future. thanks again |
| 16:48.07 | [TK]D-Fender | nsgn, You really should validate before you add a member.... |
| 16:48.15 | nsgn | that was the config change ;) |
| 16:48.22 | nsgn | lesson learned |
| 16:48.29 | [TK]D-Fender | Excellent |
| 16:48.56 | nsgn | just ticked me a bit that the user kept keying in dumb things over and over to try and log in instead of rinigng for help when they realized they took the whole queue down |
| 16:49.33 | nsgn | also discovered i can kick queue agents from FOP2 via gui. convenient! |
| 16:53.05 | nsgn | i'm out. thx again |
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| 17:19.17 | savant42 | Hey everyone, I've got a strange one for you. I'm looking at some experiments with a BAUDOT tdd and interfacing with either asterisk or linux. I understand that the tdd code is very alpha. Is the support still limited to zaptel/dahdi devices? |
| 17:22.21 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:22.30 | MichaelB8225 | good morning |
| 17:22.34 | MichaelB8225 | anyone online able to help a noob? |
| 17:23.15 | _Corey_ | savant42: I've heard of it working via SIP... apparently the baud rate is extremely low |
| 17:23.46 | savant42 | yeah, although with 45-50 bps, speed isn't really much of a concern. I get reliable 9600 connections over sip |
| 17:24.09 | savant42 | not with baudot, though. Trying to make an EAGI that will pass text from the TDD to an external application |
| 17:24.35 | MichaelB8225 | I've got Asterisk 1.8 setup on Ubuntu 12.04, its runnning, and I followed the steps here https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality |
| 17:24.51 | MichaelB8225 | but when I try to connect with a softphone (3cx free) it just says "No matching peer found" |
| 17:25.03 | savant42 | actually, if there's a gnu util that will convert raw audio containing baudot to text I could just call it externally; real time comms isn't really a concern |
| 17:25.24 | _Corey_ | savant42: There's a SIP softphone somewhere with TDD support... I can't remember off the top of my head which it was, but I suspect you're going to be fine with your application |
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| 17:26.36 | savant42 | thanks _Corey_, I'll dig deeper. I'm doing some demos at the defcon security conference; I've already got nmap by phone working using dtmf tones to enter IP addresses and festival to reads the results. I'd love if I could use a tdd to enter fqdn domain names for nmap |
| 17:26.44 | [TK]D-Fender | MichaelB8225, pastebin your sip config (masking only passwords) and the SIP DEBUG from * CLI for your attempt |
| 17:26.46 | [TK]D-Fender | ~pb |
| 17:26.46 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:26.47 | [TK]D-Fender | ^^^^^ |
| 17:27.38 | _Corey_ | savant42: lol, you could always speech-enable the thing w/lumenvox... |
| 17:27.54 | MichaelB8225 | ok, something else just came up, so I will do that in a few minutes. Thank you, Fender |
| 17:28.07 | savant42 | I was actually thinking about that, but I'd hate to inadvertantly scan the white house if I mumble ;) |
| 17:30.28 | JamKo | Am I able to specify multiple IPs for the ACL Sip Permit, for a peer, which are not successive? I have tried multiple formats, but it does not appear to work, without specifying subnet bits. |
| 17:32.41 | [TK]D-Fender | JamKo, you have to enter each as hosts or similar. |
| 17:32.49 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:34.47 | JamKo | [TK]D-Fender: So a separate peer entry in sip.conf for each? |
| 17:35.17 | [TK]D-Fender | No, multiple permit= lines |
| 17:35.24 | slav3_kitten | huzzah, hotel loaned me 2 network cables |
| 17:35.27 | [TK]D-Fender | per host if it isn't a maskable range |
| 17:37.28 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:40.01 | JamKo | [TK]D-Fender: Thanks.. I had read that asterisk only uses the last permit rule it reads, top down. |
| 17:40.09 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 17:40.35 | [TK]D-Fender | JamKo, IIRC you should be able to have multiple. just remember to put them after your deny's |
| 17:41.07 | JamKo | Thanks I'll try it. |
| 17:43.03 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:43.37 | MichaelB8225 | Hi Fender... here is the pastebin: http://pastebin.ca/2159065 I have only included the sections I have added to the conf files. The rest is out-of-the-box |
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| 17:50.27 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:52.17 | [TK]D-Fender | MichaelB8225, And the debug for the failed calls? |
| 17:55.35 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 17:55.46 | [TK]D-Fender | MichaelB8225, And the debug for the failed calls? |
| 17:55.59 | *** part/#asterisk savant42 (~savant42@adsl-99-59-240-169.dsl.pltn13.sbcglobal.net) |
| 17:56.16 | [TK]D-Fender | And... you might want to do something about your connection jumping up and down like a Jack Russell Terrier on a coke-binge.... |
| 17:57.09 | MichaelB8225 | it seems to have an exceedingly short idle timer. I'm not really sure what I can do about that. |
| 17:57.21 | [TK]D-Fender | ok, go get that debug... |
| 17:57.46 | MichaelB8225 | lost the command for that in my numerous disconnects. |
| 18:00.13 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.182) |
| 18:00.24 | [TK]D-Fender | "sip set debug on" |
| 18:00.28 | [TK]D-Fender | "core set verbose 10" |
| 18:01.48 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:02.48 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 18:04.42 | *** join/#asterisk Jouva (~jouva@pool-74-103-150-176.phlapa.fios.verizon.net) |
| 18:06.13 | Jouva | Ugh why can't I figure this out. I had this right before but then the server crashed and the admin never backed up the config :P What do I need other than nat=yes and qualify=yes to make voice work with an asterisk server in a data center w/ a public IP and hard phones behind a nat? |
| 18:07.09 | Jouva | I know we never used STUN or any similar setup, but I also don't recall having to do much of anything with port forwarding |
| 18:07.44 | Jouva | If it's either, it's port forwarding |
| 18:07.51 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 18:08.31 | jpsharp | You shouldn't have to do anything other than nat=yes. |
| 18:08.38 | jpsharp | and qualify=yes. |
| 18:08.54 | jpsharp | It only gets slightly more hokey if Asterisk is behind NAT. |
| 18:09.10 | [TK]D-Fender | Jouva, "cenreinvite=no", "directmedia=no" |
| 18:09.19 | [TK]D-Fender | ^^^ |
| 18:09.22 | [TK]D-Fender | careinvite* |
| 18:09.25 | Jouva | ahh need no, ok |
| 18:09.26 | [TK]D-Fender | canreinvite* |
| 18:09.26 | [TK]D-Fender | gah |
| 18:09.31 | Jouva | for reinvite |
| 18:10.07 | MichaelB8225 | Fender, sorry, can you repeat the sip debug command? |
| 18:10.19 | Jouva | AHA THANKS! |
| 18:10.31 | Jouva | it was canreinvite |
| 18:14.34 | JamKo | Where can I see the ACL for a given peer? >"sip show peer" does not give the permit list. |
| 18:15.09 | *** join/#asterisk MichaelB8225 (~MichaelB8@70.68.158.248) |
| 18:15.14 | [TK]D-Fender | MichaelB8225, <[TK]D-Fender> "sip set debug on" |
| 18:15.15 | [TK]D-Fender | <PROTECTED> |
| 18:20.09 | *** join/#asterisk ravnx (~mpalmer@dash.thecci.net) |
| 18:20.25 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:21.23 | ravnx | on 1.6.2 can i disable t38 inside the dial plan? I'm doing a simple call forward and getting RTP Read Too Short. |
| 18:21.38 | ravnx | only on one extension |
| 18:25.48 | *** join/#asterisk MichaelB8225 (46449ef8@gateway/web/freenode/ip.70.68.158.248) |
| 18:25.59 | MichaelB8225 | ok lets see if the Java chat is more stable than mirc. |
| 18:26.16 | MichaelB8225 | Fender, the SIP debug is in the same pastebin... http://pastebin.ca/2159113 |
| 18:26.37 | JamKo | Say I have a provider, which I need to setup twice for incoming calls, because for some ATAs I need to allow alternate codecs, and reinvites. How does.. |
| 18:27.10 | JamKo | asterisk decide which peer entry, the call will go through, if the 2 entries have the same originating ip address? |
| 18:28.02 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 18:28.27 | [TK]D-Fender | MichaelB8225, Contact: <sip:110@10.1.107.64:55754;rinstance=d680765f33dc3df0> |
| 18:28.38 | [TK]D-Fender | MichaelB8225, Yuo're trying to register to [110] |
| 18:28.53 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 18:28.57 | ravnx | Jamko: you could use dial prefixes and then set the trunks to only allow dial prefixes. |
| 18:29.04 | [TK]D-Fender | MichaelB8225, [michaelbuckley] <-- but this is a NAME |
| 18:29.10 | [TK]D-Fender | you are not "110". |
| 18:29.21 | [TK]D-Fender | you are "michaelbuckley" |
| 18:31.04 | MichaelB8225 | ok so in 3cx, I have the extension as 110, the ID as michaelbuckley and my password. |
| 18:32.20 | [TK]D-Fender | "110" is not a valid piece of identifying information |
| 18:32.24 | [TK]D-Fender | it is all named. |
| 18:33.09 | [TK]D-Fender | change your 3CX accordingly |
| 18:33.12 | MichaelB8225 | bingo! |
| 18:33.13 | MichaelB8225 | thank you |
| 18:33.21 | [TK]D-Fender | You're welcome |
| 18:34.12 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 18:34.32 | JamKo | ravnx: Ok but the provider would have to send the DID with a prefix, no? |
| 18:35.05 | ravnx | ohh, yes for incoming it would need a prefix. |
| 18:35.38 | [TK]D-Fender | JamKo, "type=friend" vs "type=peer" |
| 18:35.43 | ravnx | i've got openser in front of asterisk so we inject prefixes before the call is sent to ast |
| 18:37.38 | MichaelB8225 | so, correct me if I'm wrong (again, noob)... if I get someone else on a softphone, and they dial 110, then Asterisk will put that through to me? |
| 18:38.56 | [TK]D-Fender | yes |
| 18:39.11 | [TK]D-Fender | They dial a number and the dialplan processes that and calls your named SIP device |
| 18:40.25 | [TK]D-Fender | MichaelB8225, That is a good super-minimal start. Soon you'll be ready to look at things like adding VoiceMail for whn you don't answer, being able to pick up your VM's, menus, etc |
| 18:40.46 | MichaelB8225 | yeah, baby steps. |
| 18:40.46 | [TK]D-Fender | MichaelB8225, As your needs and desire to expand grow |
| 18:41.28 | MichaelB8225 | my previous employer used Mitel... my current employer prefers open source over vendor lock in... I'm no mitel expert, but this is a lot different. |
| 18:42.18 | [TK]D-Fender | MichaelB8225, * is an engine.... or more like an assortment of engine parts.... what the end product does is up to you. |
| 18:42.33 | [TK]D-Fender | MichaelB8225, For me it's a coffee timer and jukebox. |
| 18:43.13 | MichaelB8225 | yup, as I can see. I have a lot to learn. |
| 18:44.34 | ravnx | hrm.. FAXOPT(gateway)=no in the dialplan should do it, but not in 1.6.2 :( |
| 18:44.49 | JamKo | [TK]D-Fender: What are you referring to, my ACL >sip show peer question, or the 2 entries for one provider dilemma? |
| 18:45.13 | [TK]D-Fender | latter |
| 18:45.14 | Jouva | [TK]D-Fender: If you didn't get my thanks already, thanks for pointing out that I needed canreinvite to be no for NAT |
| 18:45.23 | [TK]D-Fender | Jouva, No problem |
| 18:45.30 | Jouva | I think I had it on that but that I was having other issues and set it to yes |
| 18:53.47 | JamKo | [TK]D-Fender: Thanks.. I'll have to think that over to decrypt the message. As for seeing the ACL for a peer. Is there a way to see that? "sip show peer" does not reflect the permit. |
| 18:54.55 | [TK]D-Fender | JamKo, No idea.... |
| 18:58.04 | JamKo | ok thanks. |
| 19:04.21 | pabelanger | anybody using option 66 with linksys spa942's, to pass the provisioning server address? |
| 19:09.53 | pabelanger | okay, so it looks like option 66 for the spa942 only supports TFTP |
| 19:09.54 | pabelanger | boo |
| 19:10.00 | leifmadsen | pabelanger: boooooo indeed |
| 19:10.20 | [TK]D-Fender | Really? hrm |
| 19:10.42 | [TK]D-Fender | I know you con provision off other methods, but you're seeing that 66 only points to TFTP? |
| 19:12.04 | pabelanger | Right, I think it will only get the initial config file via TFTP, then us HTTP is the file says too |
| 19:12.07 | pabelanger | testing it now |
| 19:12.44 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 19:18.05 | slav3_kitten | huh i dislike this cisco bullshit at the moment |
| 19:22.38 | [TK]D-Fender | Good. You understand that Cisco = bullshit. 11 steps to go! |
| 19:25.09 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:31.00 | slav3_kitten | hell yea, got it fixed, phone is upgraded to sip firmware and unprovisioned |
| 19:31.11 | slav3_kitten | this was a win 7 problem |
| 19:33.09 | resist0r | Good. You understand that win 7 = bullshit. 11 steps to go! |
| 19:33.12 | resist0r | heh |
| 19:34.49 | [TK]D-Fender | prepares to file suit.... |
| 19:35.13 | resist0r | cites fair use |
| 19:36.05 | slav3_kitten | looks an thinks he killed his 7911 |
| 19:36.18 | *** join/#asterisk sarsaeol_ (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 19:36.24 | slav3_kitten | i'm going to blame [TK]D-Fender on this one :P |
| 19:36.51 | slav3_kitten | oh wait the phone looks dead but the tftp is being active so can't be too dead |
| 19:37.02 | slav3_kitten | i hope |
| 19:37.28 | resist0r | is it still acquire'n firmware? / updates? |
| 19:37.32 | resist0r | I wonder |
| 19:37.55 | slav3_kitten | in theory? |
| 19:38.15 | slav3_kitten | it has nothing on it for signs of life... shit's blinking now |
| 19:38.22 | slav3_kitten | oh an it shows life again ll |
| 19:38.31 | slav3_kitten | lol* |
| 19:38.59 | slav3_kitten | huzzah, unprovisioned |
| 19:40.04 | slav3_kitten | i should email the guy an be all "hey, uhm this thing will appear dead for like 5 minutes while upgrading firmware... |
| 19:41.00 | slav3_kitten | it also eems to want a XMLDefault.cnf.xml |
| 19:43.27 | *** join/#asterisk zxvff (ident@pp4l.us) |
| 19:49.37 | *** join/#asterisk brimstone (~brimstone@unaffiliated/brimstone) |
| 19:49.42 | *** part/#asterisk brimstone (~brimstone@unaffiliated/brimstone) |
| 19:53.57 | slav3_kitten | can't concentrate, need food |
| 20:02.45 | RubyRails | anyone have a good example dial plan for saving QOS variables to the CDR? |
| 20:03.21 | RubyRails | exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS}) doesn't work for me |
| 20:03.26 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-dzmshlpkvlyponsm) |
| 20:03.37 | RubyRails | exten => h,1,Set(CDR(userfield)=1) sets my user field fine |
| 20:03.42 | RubyRails | but the former leaves it blank |
| 20:03.48 | RubyRails | (using a SQL DB) |
| 20:06.50 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-uhzafsmgrjzeccdu) |
| 20:06.59 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:17.14 | pabelanger | Yup, shitty. You can only do option 66 to a tftp server, then get the config file to provision from http for the SPA942 |
| 20:18.38 | drmessano | Yep.. I have to config devices in 2 stages |
| 20:19.45 | pabelanger | drmessano: Ya, I was trying to get away from tftp, but guess I can't for this site |
| 20:19.47 | drmessano | I made the best of it by including the firmware update in the first stage, but still |
| 20:20.03 | pabelanger | nods |
| 20:24.24 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 20:39.29 | *** join/#asterisk uskerine (~uske@58.Red-88-1-239.dynamicIP.rima-tde.net) |
| 20:39.31 | uskerine | hi |
| 20:39.41 | uskerine | Dial/Hangup |
| 20:39.52 | uskerine | what is missing here? I can hear voice from caller but not from calee |
| 20:39.55 | uskerine | ? |
| 20:40.16 | uskerine | exten => _X.,1,Verbose(1,Default Dial Call) |
| 20:40.25 | uskerine | exten => _X.,n,Dial(SIP/${EXTEN},50) |
| 20:40.34 | uskerine | exten => _X.,n,Playback(vm-nobodyavail) |
| 20:40.41 | uskerine | exten => _X.,n,Hangup() |
| 20:44.03 | *** join/#asterisk ATL_Tech (~tech3@173-165-184-121-atlanta.hfc.comcastbusiness.net) |
| 20:44.08 | ATL_Tech | Hello |
| 20:45.38 | l0st-soul | uskerine: i guess it has nothing to do with your dialplan but rather RTP problems |
| 20:45.47 | ATL_Tech | I'm having some issues with calls being dropped in 1.8.7 It only happens with outbound calls where the user has to navigate an IVR outside out system |
| 20:45.52 | l0st-soul | (ports, nat, ..) |
| 20:46.23 | ATL_Tech | like the DTMF tons are causing the call to drop |
| 20:48.08 | uskerine | l0st-soul |
| 20:48.24 | uskerine | from one phone i call another, callee can be heard but not the caller |
| 20:48.28 | _Corey_ | ATL_Tech: Do you have call progress detection enabled? |
| 20:48.33 | uskerine | from other phone it happens the opposite |
| 20:48.39 | uskerine | any tip l0st-soul? :( |
| 20:49.09 | l0st-soul | both phones are in the same lan without any nat or firewall between them ? |
| 20:49.10 | drmessano | ?sipnat |
| 20:49.14 | drmessano | ~sipnat |
| 20:49.14 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 20:49.38 | uskerine | both phones are in same lan |
| 20:49.45 | uskerine | asterisk is in different LAN |
| 20:49.56 | uskerine | but gateway redirects to proper LAN, FW is open |
| 20:50.14 | l0st-soul | no nat thus, only routing between the lans ? |
| 20:50.20 | uskerine | so there is a FW which divides phones LAN and asterisk LAN |
| 20:50.22 | uskerine | yes, routing |
| 20:50.22 | uskerine | no NAT |
| 20:50.48 | uskerine | should be something like bindport? |
| 20:50.51 | uskerine | or allowoverlap? |
| 20:50.57 | uskerine | or any other missing stuff in sip.conf? |
| 20:51.03 | l0st-soul | mh never had the problem in a lan, only had it when nat was involved |
| 20:51.22 | uskerine | maybe there is something NATing I am not really aware of |
| 20:51.24 | uskerine | i will double check |
| 20:51.39 | uskerine | maybe gateway is natting, even i think it is not |
| 20:52.04 | l0st-soul | does the asterisk see the phone as originiting from the phone's real ip? |
| 20:52.30 | l0st-soul | when you sip show peers |
| 20:53.22 | ATL_Tech | _Corey_ where is the option at in 1.8? |
| 20:53.34 | _Corey_ | ATL_Tech: chan_dahdi |
| 20:56.08 | ATL_Tech | yes it is enabled |
| 20:56.55 | *** join/#asterisk uskerine (~uske@58.Red-88-1-239.dynamicIP.rima-tde.net) |
| 20:57.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:58.10 | _Corey_ | ATL_Tech: I've had that cause what you're describing before, so I'd try turning it off |
| 20:58.23 | uskerine | l0st-soul, they are not NATed |
| 20:58.34 | uskerine | each phone has its own LAN IP |
| 20:58.41 | uskerine | different LAN from the asterisk server |
| 20:59.19 | ATL_Tech | _Corey_: what exactly does it do? |
| 21:00.13 | _Corey_ | ATL_Tech: Well, if I remember correctly it listens to analog channels to guess what's going on with the call ("RINGING", "BUSY", etc) |
| 21:01.06 | ATL_Tech | _Corey_: since I'm using a SIP trunk I guess it won't make a difference? |
| 21:01.55 | _Corey_ | ATL_Tech: Well, in that case it may be totally unrelated... but unless you've got POTS lines you wouldn't derive any benefit from it anyway, so I'd turn it off |
| 21:02.19 | _Corey_ | (the sample config specifically warns of random hangups...) |
| 21:06.23 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 21:08.27 | ThinkGNU- | I'm having trouble parking calls when the call rings in and I'm using local channels to designate which phones ring. If I pickup from one of those phones and try to park a call it won't acknowledge the key sequence. |
| 21:10.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:23.59 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 21:24.02 | ThinkGNU- | It seems as if my phone isn't allowing the DTMF signaling to park the call |
| 21:26.03 | *** part/#asterisk agisamentum (~vastina@70.99.103.122) |
| 21:26.42 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:37.04 | *** join/#asterisk d_preston215 (~chatzilla@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 21:37.09 | d_preston215 | Can I pass multiple variables via Set in a call file? |
| 21:45.43 | slav3_kitten | repeatedly head desks |
| 21:49.05 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jihxbkmxztaptqhi) |
| 21:52.11 | d_preston215 | I know. |
| 21:56.40 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 21:56.58 | trelane | is there a configuration guide for the new Mediatrix 41xx hardware/firmware and Asterisk? |
| 22:00.10 | drmessano | Not much to configure |
| 22:07.15 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 22:11.37 | *** join/#asterisk danfromuk (~IceChat77@2.27.7.133) |
| 22:23.30 | ThinkGNU- | Can anyone tell me what the best way to setup a night ringer mode would be? I can use the GotoIfTime function but I also need there to be an over ride to manually turn it off or on. |
| 22:24.13 | leifmadsen | ThinkGNU-: just set a GLOBAL() var to check for the override |
| 22:24.14 | l0st-soul | astdb? |
| 22:24.17 | leifmadsen | or that |
| 22:24.23 | leifmadsen | both are valid |
| 22:24.39 | l0st-soul | one will survive a restart |
| 22:24.39 | leifmadsen | astdb probably is the better way (loads same value as previous after restart) |
| 22:24.42 | leifmadsen | that * |
| 22:24.49 | ThinkGNU- | okay |
| 22:25.38 | ThinkGNU- | I was thinking of setting a rapid dial key to trigger a part of the dial plan to set the value. I was also thinking of setting a hint to light up the rapid dial key depending on the status of the override |
| 22:25.41 | ThinkGNU- | How does that sound? |
| 22:25.53 | l0st-soul | when i began with asterisk i used globals to setup my primary and secondary trunks |
| 22:26.01 | l0st-soul | and invert them when wanted |
| 22:26.19 | l0st-soul | after restart when i wasnt at work, they had a bad surprise |
| 22:26.59 | ThinkGNU- | yeah, we don't want anything like that 10st-soul |
| 22:27.31 | ThinkGNU- | This is my first Asterisk setup so that's why I'm trying to solicit a bit of feedback from the pros |
| 22:27.36 | l0st-soul | i'd say its doable |
| 22:27.54 | l0st-soul | i did something like that for manual night mode / day mode |
| 22:28.27 | l0st-soul | check if db value for night is on off |
| 22:28.42 | l0st-soul | and create an extension like *66 to toggle the state |
| 22:28.57 | ThinkGNU- | and were you having the value checked with every incoming call? |
| 22:29.03 | ThinkGNU- | Then branching your dialplan from there? |
| 22:29.18 | l0st-soul | (+ send a sip text message to the phone's lcd to confirm the mode has been enabled/disabled) |
| 22:29.33 | ThinkGNU- | That's a great idea I didn't think of |
| 22:30.40 | l0st-soul | or just say "activated" or "disabled" on the phone when toggling |
| 22:30.43 | *** part/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 22:31.00 | *** join/#asterisk gorilla (c8317ffd@gateway/web/freenode/ip.200.49.127.253) |
| 22:31.03 | l0st-soul | well i did it for the office's main reception number |
| 22:31.36 | l0st-soul | but you want it to be scheduled + have the manual switch ? |
| 22:31.41 | l0st-soul | we only use the manual switch |
| 22:31.52 | ThinkGNU- | I was thinking so yeah |
| 22:32.07 | l0st-soul | but i'm sure you'll figure a place in your dialplan where you can make the check |
| 22:32.32 | WIMPy | custom devstates are great for such things. |
| 22:32.33 | ThinkGNU- | I was thinking of doing an on, off, auto setting |
| 22:33.38 | l0st-soul | ultimately for the auto off/on, it could even be done in a crontab |
| 22:33.38 | ThinkGNU- | If it's on auto I'll have the rapid dial blink, if it's on "on" it will be lit up and if it's off the rapid dial button won't be lit |
| 22:33.59 | l0st-soul | sending commands to the console to database put blah blah |
| 22:34.08 | ThinkGNU- | that's true |
| 22:34.23 | ThinkGNU- | I really like how flexible asterisk is but sometimes it overwhelms me |
| 22:34.23 | l0st-soul | but i'm sure its not even needed |
| 22:34.46 | ThinkGNU- | plus, like I say this is my first system |
| 22:34.46 | l0st-soul | are you using a regular extensions.conf or .ael ? |
| 22:34.53 | Guest61401 | Hi all, I would test a Digium E1 Card in standalone configuration. Is possible ? if yes, which configuration I can use ? |
| 22:34.59 | ThinkGNU- | the regular extensions.conf |
| 22:35.01 | slav3_kitten | ThinkGNU-, i feel your pain |
| 22:35.01 | l0st-soul | i recently switched to ael and OH GOD it's so good |
| 22:35.10 | slav3_kitten | i' sitting here trying to figure out how to provision 2 phones :/ |
| 22:35.22 | l0st-soul | if you're a bit used to basic programmation |
| 22:35.22 | slav3_kitten | stupid cisco phones |
| 22:35.27 | WIMPy | Guest61401: What do you want to test? |
| 22:35.34 | l0st-soul | its so much clearer to write and modify dialplans |
| 22:36.12 | ThinkGNU- | 10st-soul I'm just getting myself into programming. I'm getting a bit quicker with emacs and all of it's commands too |
| 22:36.41 | l0st-soul | but you have a bit of understading of structures like |
| 22:36.48 | l0st-soul | if (.....) { |
| 22:36.51 | ThinkGNU- | yes |
| 22:36.53 | l0st-soul | <PROTECTED> |
| 22:36.54 | l0st-soul | } |
| 22:36.59 | l0st-soul | etc |
| 22:37.10 | ThinkGNU- | I've done just enough of that to get things working (after many typos usually) |
| 22:37.14 | Guest61401 | WIMPy: I would call across a channel and receive call on another. |
| 22:37.29 | l0st-soul | i find its such an improvement in clarity compared to all those exten=> lines |
| 22:37.38 | ThinkGNU- | I can see how that could be |
| 22:37.51 | WIMPy | Guest61401: You can build a cross over cable if you have a multi port card. |
| 22:38.15 | ThinkGNU- | Well I wish I could say I was setting up a system for just eight phones or something small like that but I'm setting up for about 25 phones actually. It's just a bit more than I'd like to do for my first system. I'll get it though. |
| 22:38.35 | ThinkGNU- | I'd rather go with * than something completely proprietary that I have to call a company in to adjust for us |
| 22:39.23 | l0st-soul | i ran into thesame question 2 or 3 years ago |
| 22:39.30 | l0st-soul | with 50 phones |
| 22:39.51 | citywok | i learned * by building the phone system our 125 seat call center uses. lol |
| 22:39.56 | ThinkGNU- | what were your biggest challenges getting started with your first setup? |
| 22:40.02 | l0st-soul | i felt i could do it myself but the boss wasnt reassured, so we finally had the * installed by an external company |
| 22:40.12 | l0st-soul | then i took it over |
| 22:40.42 | citywok | ThinkGNU-: well, i didn't know much of * so i had to learn Queues, autodialing, and every other feature our call center needed. And i only had 6 months to figure it out and make it work, and integrate it with our existing systems. |
| 22:40.53 | l0st-soul | and now i reinstalled a new * and rewrote the dialplan completely |
| 22:41.43 | ThinkGNU- | I'm sure learning a lot here. I'm thankful for the documentation and the willingness of folks here to graciously answer my questions |
| 22:41.55 | ThinkGNU- | I'm closing in on having things done but it's just taking some time |
| 22:42.29 | l0st-soul | i had a lot of trial and errors |
| 22:42.36 | l0st-soul | a lot of debug |
| 22:42.37 | citywok | ThinkGNU-: that's because we were all noobs at one time too. |
| 22:42.42 | l0st-soul | a lot of googling |
| 22:42.47 | ThinkGNU- | I'm glad it's not just me |
| 22:42.52 | l0st-soul | and when it works the way you want, whooo |
| 22:43.12 | ThinkGNU- | Here's a quick question: How do I make the CallerID value from an originating line stick when that call is transferred? |
| 22:43.45 | l0st-soul | i'm interrested in doing that as well, now that you mention it |
| 22:44.10 | ThinkGNU- | citywok? Do you know the answer? |
| 22:44.28 | l0st-soul | you're talking of an unattended transfer i suppose |
| 22:45.11 | l0st-soul | or by transfered, you mean deviated/forwarded automatically by a sip phone |
| 22:45.32 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:45.49 | ThinkGNU- | yes |
| 22:46.01 | ThinkGNU- | I think we're talking about the same thing. |
| 22:47.19 | l0st-soul | if i setup a forwarding on my snom sip phone, to another sip phone, i think the callerid is following correctly |
| 22:47.52 | Guest61401 | WIMPy: I build cross over cable but my card has only one port. Someone told me MFC/R2 support calls on loopback cable. is true ? |
| 22:47.53 | l0st-soul | but if i have someone on the phone, and i transfer that call to a colleague, he sees my number |
| 22:48.10 | l0st-soul | or maybe i'm wrong now,i dont know anymore .. |
| 22:48.54 | WIMPy | Guest61401: You can build a loopback plug to test the port, but you can not send calls on only one port. |
| 22:49.05 | l0st-soul | ThinkGNU-: but the thing that is sure and apparently has no solution, is an Attended transfer (got someone on the phone, i dial a colleague on 2nd line telling him i'm transfering him the call |
| 22:49.18 | l0st-soul | in that case he'll still see my Callerid on his screen |
| 22:49.32 | ThinkGNU- | That makes sense because you're able to tell him who you have on the line |
| 22:49.56 | ThinkGNU- | If I do a blind transfer from one phone to another it gives the receiving phone the calling phone's CallerID info. |
| 22:50.11 | l0st-soul | yeah but i find it annoying he keeps my number on his screen the whole conversation |
| 22:50.23 | l0st-soul | but i understand callerid is something setup at call initiation, not during conversation |
| 22:51.02 | l0st-soul | and once the transfered call is answered by phone2, how many channels do you see in * ? |
| 22:51.08 | l0st-soul | i mean core show channels |
| 22:51.11 | l0st-soul | 2 or 4 ? |
| 22:51.18 | ThinkGNU- | I haven't checked that |
| 22:51.22 | ThinkGNU- | I'll have to look that over |
| 22:51.34 | l0st-soul | maybe phone1 is really creating a 2nd call |
| 22:51.43 | l0st-soul | just a guess |
| 22:51.49 | ThinkGNU- | That's very possible. |
| 22:52.15 | ThinkGNU- | I don't know a way to make it not do that if we're using the phone's transfer button rather than a DTMF signal to trigger a transfer. |
| 22:53.16 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 22:53.26 | Guest61401 | WIMPy: thank you. Another question, on PRI T1, I need incoming calls on channels 1-10 go to context from-x1 and channels 11-23 goto context from-x2. is possible ? |
| 22:53.32 | l0st-soul | maybe a complex dialplan that is checking if there is another channel existing with the same phone number |
| 22:53.50 | l0st-soul | and spoofing the callerid of the call with the caller's callerid of the other call |
| 22:53.57 | l0st-soul | before dialing the phone |
| 22:54.01 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 22:54.03 | WIMPy | Guest61401: Yes, but what kind of sense could that make? |
| 22:54.12 | l0st-soul | but that looks like madness |
| 22:54.27 | ThinkGNU- | 10st-soul, I agree |
| 22:55.12 | l0st-soul | i think i'll just live with the 'problem' |
| 22:55.13 | l0st-soul | :) |
| 22:55.26 | ThinkGNU- | Yeah, it's not that bad really. It's just what it is. |
| 22:55.50 | ThinkGNU- | Thanks for the info and reasurance here guys. I'll be on again but I need to get going now. |
| 22:55.51 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 22:56.02 | l0st-soul | i need to sleep also |
| 22:56.03 | l0st-soul | bye folks |
| 22:56.09 | ThinkGNU- | later |
| 22:56.11 | *** part/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 22:56.43 | slav3_kitten | biggest pain in the ass i've ever run into |
| 22:57.25 | Guest61401 | WIMPy: yes, sure; and define groups of outbound channels make sense ? |
| 22:57.46 | WIMPy | Guest61401: No that does not make sense on a PRI. |
| 22:57.48 | slav3_kitten | so i want to setup this netbook in this hotel room to be a tftpd and dhcpd |
| 22:58.17 | slav3_kitten | you'd think it was easy but the wifi here blows hard, and i keep getting errors apt-getting |
| 22:59.13 | *** join/#asterisk jackmcbarn (47c77dd2@gateway/web/freenode/ip.71.199.125.210) |
| 23:00.17 | Guest61401 | WIMPy: thanks, and sorry for the silly questions, but I never had a T1. |
| 23:00.21 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 23:01.24 | nny | ok confirming something here. Having provider outbound issues (provider uses broadsoft). Getting abnormal 401s from them, see http://i.imgur.com/VYO0j.png |
| 23:03.12 | nny | sorry broadvoice* lol |
| 23:03.38 | sarsaeol_ | you're right broadvoice uses broadsoft |
| 23:04.13 | jackmcbarn | on my DAHDI channels, my phone keeps getting "missed calls" after regular calls hang up, and when asterisk restarts. the caller id of the "missed calls" is the caller id of the line being called. the phone doesn't actually ring when this happens. |
| 23:06.23 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-14-184.bois.qwest.net) |
| 23:07.01 | JamKo | If I have a UA set to canreinvite=no , with a provider set to canreinvite=yes , is a reinvite possible if the provider initiates it? |
| 23:11.52 | [TK]D-Fender | no |
| 23:11.58 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1828.pool.mediaWays.net) |
| 23:15.56 | slav3_kitten | question: do other voip phones pull their config from a tftp server? |
| 23:17.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:18.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:20.54 | nny | [TK]D-Fender: wrestling with a broadvoice/works/soft reseller and asterisk config. I have http://i.imgur.com/VYO0j.png happening on the wire, but only see 102/103 over and over in sip debug. I am thinking this is a config issue, but their support hasn't provided the most accurate config examples. http://pastebin.com/V193CrxT thoughts? |
| 23:22.01 | nny | if I use their outbound dial using exten => _1xxxxxxxxxx,n,SipAddHeader(P-Asserted-Identity: <sip:7782185795>), it fails with Forbidden. If I just dial it, I get the 102/103/401 dance. |
| 23:22.20 | nny | note i have mangled the user name etc, but probably not very well. It's not easy to do with so much info |
| 23:28.42 | nny | anyone else wanna take a stab at that? I really hate dealing with this provider, it's the same round of configuration oddities everytime |
| 23:29.03 | slav3_kitten | nny, wish i chould help |
| 23:29.16 | slav3_kitten | i'm currently beating my brains out trying to figure out how to configure these cisco phones |
| 23:29.32 | slav3_kitten | and general not having a fucking clue what i'm doing as i'm not even half through the book |
| 23:30.46 | nny | what kind of ciscos? |
| 23:30.59 | slav3_kitten | 7911 and 7940 at the moment |
| 23:31.06 | slav3_kitten | 7921 sometime in july |
| 23:31.10 | nny | i have some tftp configs for a 7940 |
| 23:31.18 | slav3_kitten | ?! seriously |
| 23:31.21 | nny | yeah one sec |
| 23:32.18 | slav3_kitten | brb need to look into making dinner, stomach is making feed me noises |
| 23:35.18 | nny | slav3_kitten: http://pastebin.com/KmH4FA6E + you'll need the SIP firmware in the tftp directory. There's two file examples there, SIP<MAC>.cfg and SIPDefault.cnf. The phones only look for the file "SIPDefault.cnf" so you can't change the name. |
| 23:35.18 | nny | After that you have to unlock the phone from the LCD and enter the TFTP server info (or Option 66) and reboot |
| 23:35.46 | jackmcbarn | how can i view detailed dahdi debugging info? |
| 23:36.07 | carrar | nny, besure to note that is only for 7940/7960 |
| 23:37.17 | jackmcbarn | i set debug to 1 in the dahdi kernel module, but i dont know where it goes |
| 23:37.51 | slav3_kitten | carrar, how does a 11 differ aside from config changes to the sip<mac> file? |
| 23:38.12 | carrar | I've not used a 7911, but I'm guessing it's a XML config file |
| 23:38.36 | carrar | like the 7941/61/70/75 |
| 23:38.52 | nny | aye i did |
| 23:39.10 | carrar | and does the 7921 even do SIP? |
| 23:39.21 | nny | slav3_kitten: I can upload the sip loads I have for the 7940 if you can't find them as well |
| 23:40.00 | nny | /srv/tftpboot/P003-8-12-00.bin /srv/tftpboot/P003-8-12-00.sbn /srv/tftpboot/P0S3-8-12-00.loads /srv/tftpboot/P0S3-8-12-00.sb2 |
| 23:40.02 | slav3_kitten | 7921 is sccp only but i found some info about that |
| 23:40.04 | carrar | might need to use the skinny driver |
| 23:40.09 | nny | slav3_kitten: are the four files I have |
| 23:40.25 | slav3_kitten | i got sip firmware for the 11 and 60 |
| 23:40.33 | nny | slav3_kitten: you may want to consider SPA50X phones from cisco, they're ~ 100 bucks or so a piece for the 4 line phones and a thousand times easier to configure |
| 23:40.56 | slav3_kitten | got those flashed over to sip, now fighing my ghetto distro on the netbook to do tftpd properly |
| 23:41.08 | nny | slav3_kitten: can't do tftp on the pbx? |
| 23:41.25 | WIMPy | But they have the old Sipura shit. |
| 23:41.46 | slav3_kitten | nny, i have no pbx yet, i'm learning it via my netbook, a switch, and some cisco phones i have |
| 23:42.56 | slav3_kitten | i'm also sitting in a hotel room 600 miles away from home with power cords run all over because my power strip went missing along with 2 patch cables |
| 23:44.30 | slav3_kitten | the fail train is real, and i have a rail pass for it lol |
| 23:45.20 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 23:46.56 | nny | lol |
| 23:47.25 | nny | anyone here wanna take a stab at my provider woes? |
| 23:47.30 | nny | the provider isn't very helpful |
| 23:48.00 | sawgood | SIP trunk? |
| 23:48.04 | nny | sawgood: aye |
| 23:48.10 | nny | sawgood: broadvoice based one afaik |
| 23:48.16 | nny | sawgood: it's a reseller |
| 23:48.18 | sawgood | USA numbers? |
| 23:48.20 | nny | sawgood: aye |
| 23:48.24 | slav3_kitten | so yea... do other phone like polycom yank their cfg off of tftp? |
| 23:48.34 | nny | slav3_kitten: they dont' have to but they can |
| 23:48.45 | slav3_kitten | whereas cisco has to |
| 23:48.48 | slav3_kitten | i hate cisco so much |
| 23:48.52 | nny | slav3_kitten: a lot of phones just have a web ui and the optiob |
| 23:48.57 | nny | slav3_kitten: the new ciscos are way better |
| 23:49.11 | nny | slav3_kitten: the SPA stuff, the 79xx series and SIP is a cludge imho |
| 23:49.33 | slav3_kitten | eh it's a cludge i had and can afford |
| 23:49.49 | sawgood | The SPA 5xx series are cleaner to work with |
| 23:49.49 | nny | slav3_kitten: ahh well, if you can get a tftpserver up and those files on it, you should be good |
| 23:49.53 | slav3_kitten | also i'm really fond of that really stupid message waiting/call indicator light pipe |
| 23:50.13 | JamKo | [TK]D-Fender: thanks. |
| 23:50.22 | nny | slav3_kitten: the SPA has very a similar design, but yeah, tftpserver first |
| 23:50.22 | slav3_kitten | then again i've been googling ways to have a strobe light sip ringing |
| 23:51.15 | slav3_kitten | actually i'd give my soul to have sip activated dry contacts for ringing to run a revolving light i've had since i was a kid |
| 23:51.20 | slav3_kitten | if it didn't get lost in the move |
| 23:51.36 | carrar | slav3_kitten, super easy with a ATA |
| 23:51.48 | carrar | radio shack even sells a strober |
| 23:51.54 | carrar | analog which would plug into the ATA |
| 23:52.04 | slav3_kitten | no way to have it pure sip no ata cheaply? |
| 23:52.26 | drmessano | What does the SIP plug into? |
| 23:52.32 | slav3_kitten | ethernet |
| 23:52.57 | slav3_kitten | poe powered ethernet, without me hacking apart something to make it look clean |
| 23:53.17 | *** join/#asterisk singler (~singler@beta.kirneh.eu) |
| 23:53.45 | drmessano | Maybe I asked that wrong.. You want to do it with SIP, you need an interface.. You need a box with a SIP stack that's going to give provide an analog output of some sort. You can R&D something or just use a cheap ATA |
| 23:54.19 | carrar | slav3_kitten, http://www.arktel.com/catalog/p-647-sip-based-strobe-light.aspx |
| 23:54.22 | carrar | be free |
| 23:54.44 | drmessano | Thats not too bad.. $3.98 |
| 23:54.45 | carrar | I'd like to introduce you to google |
| 23:54.49 | drmessano | Wait.. |
| 23:54.59 | drmessano | :) |
| 23:55.12 | carrar | now |
| 23:55.19 | carrar | back to the ATA and radioshack strobe like :) |
| 23:55.25 | carrar | lite |
| 23:55.36 | carrar | heh |
| 23:55.37 | drmessano | Does RS even make the old Fone Flasher 2 anymore? |
| 23:55.43 | carrar | not sure |
| 23:55.47 | drmessano | The one with a receptacel, no light |
| 23:55.49 | carrar | but they are still avail by others |
| 23:55.53 | drmessano | err receptacle |
| 23:56.15 | slav3_kitten | drmessano, i've got an old diy analog phone alerter |
| 23:56.33 | drmessano | Ok, so even more of a case for an ATA |
| 23:58.23 | slav3_kitten | hmm |
| 23:58.35 | slav3_kitten | i'll think about it, not the most pressing issue |
| 23:58.54 | slav3_kitten | need to finish the book learn asterisk like the back of my hand, and play with it |
