| 00:00.08 | wswartz | My Verizon 3G phone on the other hand... |
| 00:00.47 | WIMPy | tunnels are great as well. |
| 00:01.05 | *** join/#asterisk qakhan (~qakhan@202.142.158.93) |
| 00:01.14 | qakhan | WIMPy you there? |
| 00:01.32 | WIMPy | yes |
| 00:08.45 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:11.22 | Ice_Strike | WIMPy what voip provider do u use |
| 00:12.31 | WIMPy | I've got accounts from sipgate and 1&1. But I rarely use them. |
| 00:15.05 | wswartz | Am I supposed to have to setup extensions.conf just for local users to be able to call each other? |
| 00:15.17 | WIMPy | yes |
| 00:15.24 | wswartz | Okay, just checking. |
| 00:15.27 | WIMPy | You need it to make any call. |
| 00:15.44 | wswartz | Well isn't the phone supposed to dial outside servers directly? |
| 00:15.52 | wswartz | I've read that SIP is not a proxy. |
| 00:16.11 | wswartz | And my understanding is that unless a proxy is used, the phone dials the outside server directly. |
| 00:16.30 | wswartz | But SIP in itself really confuses me. |
| 00:16.50 | WIMPy | That depends on your phone and its configuration. |
| 00:17.33 | wswartz | Do outbound calls to external servers generally go through my server? |
| 00:18.10 | WIMPy | That depends on your phone, its configuration and how you dial. |
| 00:18.39 | wswartz | Is there an accepted way to go about this? |
| 00:18.58 | wswartz | It seems like a lot of SIP is up in the air. |
| 00:19.44 | WIMPy | Pollution? :-) |
| 00:20.53 | wswartz | So is it perfectly acceptable to have the phone contact my server only for its own extensions? |
| 00:21.08 | WIMPy | Sure |
| 00:21.26 | wswartz | I wish there was a generally accepted way. |
| 00:21.26 | WIMPy | Some phones can also make ENUM queries. |
| 00:21.57 | WIMPy | By now we're happy if anything works at all. |
| 00:23.24 | wswartz | By now? |
| 00:23.52 | WIMPy | todays technology |
| 00:24.19 | wswartz | Well IPv6 can simplify some things. |
| 00:24.50 | WIMPy | For initial setup, yes. |
| 00:32.39 | huntsville-370z | why |
| 00:39.41 | wswartz | Why what? |
| 00:40.02 | *** join/#asterisk jeffspeff (~Jeff@173.216.213.80) |
| 00:42.22 | jeffspeff | does does 1.8.11 support multiple parking contexts? |
| 00:48.28 | huntsville-370z | I just picked my pinkey toenail and tried to split it in half in my hand after i sharded it off, but the calcium micro structures is preventing it from being split |
| 00:48.32 | huntsville-370z | it has a strong biological bond |
| 00:56.22 | *** join/#asterisk jeffspeff (~Jeff@173.216.213.80) |
| 01:02.52 | huntsville-370z | Phil Swift here, with Flex Seal. the easy way to coat, seal, and stop leaks before they happen. It's actually Liquid Rubber in a can....... After a day on the water, the inside is completely dry... don't call a repair man, that'll "kawst" you thOWsands........... |
| 01:17.31 | jeffspeff | does does 1.8.11 support multiple parking contexts? |
| 01:17.43 | *** join/#asterisk bytemaster (~retretret@host81-150-217-167.in-addr.btopenworld.com) |
| 01:23.38 | *** join/#asterisk engrxyz (~tyrtyrty@cpc3-basl7-0-0-cust252.basl.cable.virginmedia.com) |
| 01:34.27 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
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| 02:06.51 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 02:09.59 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 02:19.16 | wswartz | What's the whole point of subscriptions and publishing? |
| 02:22.00 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 02:38.49 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 02:48.40 | wswartz | Actually, is there a way I can get Asterisk to act as a proxy and pass calls through? |
| 02:58.25 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@97-112-228-116.chyn.qwest.net) |
| 03:07.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:18.24 | *** join/#asterisk trung (~trung@ec2-23-23-179-94.compute-1.amazonaws.com) |
| 03:18.37 | trung | HAHAH HELLO ALL! |
| 03:18.57 | trung | I thought it was asterisk was asterix for the last 5 years |
| 03:19.27 | trung | sorry had to share that.. i was wondering why #asterix and ##asterix was a ghost town |
| 03:19.53 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 03:20.39 | trung | Anyhow, a question, is there some off the shelf hardware i can which has telephony interfaces to connect to legacy phones for the purposes of redirecting unknown callers to a mailbox and not ringing the traditional phone? |
| 03:21.22 | trung | I mean i could run up an computer with a telephony interface card, but prefer something which is more 'turn on, turn off, administer via web & has ssh access' |
| 03:34.48 | ectospasm | trung: Digium sells appliances like that. |
| 03:48.38 | trung | ectospasm, is it the g100? |
| 03:49.50 | ectospasm | no, you need analog telephony, right? |
| 03:49.54 | trung | yes |
| 03:50.11 | ectospasm | G100 and G200 are digital (T1/E1) only |
| 03:50.30 | trung | oh! |
| 03:50.34 | ectospasm | you could probably get an AA65 with Switchvox on it. |
| 03:50.37 | trung | as you can see. i am very unfamiliar |
| 03:50.52 | trung | whats switchvox anyway? |
| 03:50.56 | ectospasm | ...throw in an AEX800 or whatever, with 2X400M |
| 03:51.22 | ectospasm | trung: it's a GUI that totally encapsulates Asterisk. Should be able to set it up within 30min tops. |
| 03:51.33 | trung | i see |
| 03:52.04 | ectospasm | disclaimer: I work for Digium (makers of Switchvox, and primary sponsor of the Asterisk community) |
| 03:52.20 | trung | ectospasm: that's fine. i am having difficulty on the website |
| 03:52.32 | trung | if i search AA65, the first hit should be the product! |
| 03:52.40 | trung | instead i get forum posts, which is honorable, but not helpful |
| 03:52.48 | trung | bah, i dunno.. heh just my winge |
| 03:53.13 | trung | wow, $3,000! |
| 03:53.38 | ectospasm | http://www1.digium.com/en/products/switchvox/systems/switchvox-hardware |
| 03:53.44 | trung | this is just for my home residence, because i am annoyed by random callers. can you recommend me anything else? |
| 03:54.33 | ectospasm | trung: if you want PNP? No, I can't suggest anything else. If you're willing to get your hands dirty, open source Asterisk + AEX404 is the way to go. |
| 03:54.52 | ectospasm | ...for what you want it shouldn't be too difficult to set up. |
| 03:56.13 | trung | ectospasm: that's fine, i just wanted to know what the current pricing point for PNP is.. i don't mind getting my hands dirty |
| 03:57.43 | ectospasm | http://store.digium.com/productview.php?product_code=1AEX404EF << this is the highest price you'll pay. Resellers have it for far cheaper. |
| 03:58.16 | trung | ectospasm: why would you sell it higher than resellers? |
| 03:58.20 | ectospasm | ...just have a decent box to put it in, install your favorite Linux distro, install the drivers and set up the card (Digium Install support can help with this) |
| 03:58.28 | ectospasm | trung: they get volume discounts |
| 03:58.35 | ectospasm | and we don't want to undercut our resellers. |
| 03:58.41 | trung | ahh, they take on the risk |
| 03:58.46 | trung | sounds reasonable |
| 04:00.59 | trung | wow the switchvox intro youtube is great, inspiring, informative, exciting |
| 04:12.51 | ectospasm | trung: watch out, don't let marketing get their hooks into you! |
| 04:12.55 | ectospasm | :-P |
| 04:13.24 | trung | haha i know.. i just wanted yo comment it was well made |
| 04:13.33 | ectospasm | Although Switchvox is the best option for most people that just want a minimum of futzing with it |
| 04:13.35 | trung | s/yo/to/ |
| 04:13.56 | trung | wow thanks infobot bot |
| 04:14.04 | ectospasm | remember that any GUI encapsulates away most of the flexibility. |
| 04:14.37 | trung | yeah. i love a good CLI exterience |
| 04:16.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:17.24 | ectospasm | trung: then find a local reseller, buy the card as cheap as possible, and have all the fun setting it up |
| 04:18.54 | trung | a local seller means paying australian dollars = $$$ |
| 04:18.56 | trung | :P |
| 04:21.50 | ectospasm | we do have Australian resellers. I think Australian Technology Partnerships is our distributor there. |
| 04:21.58 | ectospasm | ...but I may be wrong. |
| 04:25.52 | trung | PNP offerings look great!!! *sigh* |
| 04:26.31 | trung | i woundn't be able to get that past the minister of finance at home just to not answer 'private numbers' haha |
| 04:28.49 | wswartz | My Google Voice calls keep dying after thirty seconds. |
| 04:29.09 | ectospasm | wswartz: any hint of the problem in the Asterisk CLI? |
| 04:29.30 | wswartz | Nupe. |
| 04:29.32 | ectospasm | trung: and CID on analog can be tricky to set up. |
| 04:30.09 | ectospasm | ...especially for international customers. I don't remember having a problem with an Australian customer, but YMMV |
| 04:33.05 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 04:34.28 | trung | ectospasm: oh really? my crappy analog home phone displays CID, i'd thought it'll be easy |
| 04:35.34 | trung | interesting.. if i get international calls from family, i'd like to give them an opportunity to enter in a password for the call to go past the voice mailbox and just ring the house phone |
| 04:35.42 | ectospasm | trung: you probably use the same thing the British do, so it's not so bad. I have a lot of middle eastern customers that just can't get it working natively. |
| 04:36.14 | trung | ectospasm: i wouldn't be suprised if we do the same things the brits do! thanks for the heads up |
| 04:41.56 | *** join/#asterisk mintos (~mvaliyav@114.143.163.99) |
| 04:42.16 | mzb | trung: I'm .au and I don't get unknown callers |
| 04:42.32 | mzb | I have two solutions: |
| 04:43.21 | mzb | 1) a recording that is played for callers without CLID => "This call may be recorded for quality assurance purposes." (in accordance with Australian law) |
| 04:43.40 | mzb | (this announcement scares off 99.9% of the telemarketers) |
| 04:44.05 | mzb | 2) a CallerID lookup to ReverseAustralia |
| 04:44.06 | wswartz | Actually, my Google Voice calls seem to be getting dropped because of failure to reply to something called a critical packet. |
| 04:45.16 | mzb | checked your firewall? |
| 04:46.09 | ectospasm | wswartz: who is failing to reply, GV or Asterisk? |
| 04:47.22 | wswartz | HEADSUP: I'm gonna paste a couple lines. |
| 04:47.35 | wswartz | [May 12 21:33:21] WARNING[31622] chan_sip.c: Retransmission timeout reached on transmission 4OVdQkI5I3P6-bNuzEvZb0QGqK99GSzH-S for seqno 27868 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 04:47.35 | wswartz | Packet timed out after 32003ms with no response |
| 04:47.43 | ectospasm | ~pastebin |
| 04:47.43 | 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. |
| 04:47.58 | wswartz | Whoops. I was told three lines or less is cool. |
| 04:48.06 | ectospasm | no, not really |
| 04:48.21 | wswartz | Understood, and I apologize. |
| 04:48.41 | ectospasm | wswartz: so Asterisk has transmitted to GV the maximum number of times, and isn't getting the response it expects. |
| 04:48.56 | wswartz | I brought down iptables and it still persisted. |
| 04:49.48 | wswartz | I can still have a conversation during this time. |
| 04:50.08 | wswartz | Two-way communication gets established and then just gets cut off. |
| 04:50.36 | ectospasm | is there a standalone firewall between the Asterisk machine and GV? |
| 04:51.36 | wswartz | The server is a VPS. |
| 04:52.00 | wswartz | Maybe it's using IPv6. |
| 04:52.12 | wswartz | talk.google.com has AAAA. |
| 04:52.50 | ectospasm | ...there's something in between not xmitting Asterisk's packets to GV |
| 04:53.03 | ectospasm | ...or GV is refusing to respond, thus Asterisk tears down the channel. |
| 04:53.24 | ectospasm | ...what version of Asterisk is this? |
| 04:54.00 | wswartz | 10.4.0 |
| 04:54.15 | ectospasm | any reason you're using 10 and not 1.8? |
| 04:54.28 | wswartz | Because it's what Gentoo gave me. |
| 04:54.36 | ectospasm | meh, Gentoo. |
| 04:54.42 | wswartz | Why is 10 bad? |
| 04:54.54 | ectospasm | 10 is a standard release. It will be EOL before 1.8 is. |
| 04:55.03 | wswartz | I'm fine with that. |
| 04:55.24 | ectospasm | ...I really only recommend 10 if it has features that 1.8 lacks, and you need those features. |
| 04:55.35 | wswartz | Not until Opus is merged. |
| 04:55.52 | wswartz | So Asterisk will tear down the channel even though a conversation is happening. |
| 04:56.34 | ectospasm | wswartz: because it's not getting a response to some packet. Unfortunately I know zero about the protocol GV uses, so I can't be much help. |
| 04:56.44 | wswartz | XMPP |
| 04:57.03 | ectospasm | yep, don't know anything about the underlying jabber protocol, either. |
| 04:59.05 | ectospasm | It looks like wireshark can decode Jabber, but I don't know if that includes audio streams. |
| 05:00.59 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 05:13.01 | wswartz | It's an Android problem of some sort. |
| 05:13.20 | wswartz | Works fine on Linphone. |
| 05:20.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 05:47.47 | wswartz | Is there an easy way to get Asterisk to act as a dumb proxy? |
| 05:48.11 | wswartz | So that if I dial "echo@iptel.org", it connects to iptel.org. |
| 05:50.45 | ectospasm | that would be enabled in your endpoint, if it accepts URL dialing. |
| 05:50.55 | ectospasm | ...that wouldn't involve Asterisk at all. |
| 05:51.38 | ectospasm | ...or, if you wanted to define a short code for echo@iptel.org, just have Dial(SIP/echo@iptel.org...) |
| 05:56.55 | wswartz | Okay so traditionally, a SIP phone does not contact its registered server to place an outbound call. |
| 05:57.06 | wswartz | I'm just trying to understand the culture of how this works. |
| 06:05.52 | ectospasm | wswartz: correct, a SIP phone only registers with a proxy in order to tell the proxy where it is so the proxy knows where to send the endpoint calls. |
| 06:06.30 | ectospasm | ...but most endpoints just send outbound calls to the proxy, and let the proxy decide how/where to send them. |
| 06:09.37 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 06:14.55 | wswartz | So this proxy is part of a usual setup, it's just not Asterisk's job? |
| 06:18.52 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.241) |
| 06:32.48 | vader-- | Can any of you guys recommend a good/economical hosted pbx for a two phone office? They currently have polycom sip phones, but they are paying $200 for two phones. |
| 06:39.45 | trung | mzb: for your first solution is that easy to do with asterisK? |
| 06:40.02 | trung | mzb: how do u do a CID lookup if it's private? |
| 06:44.34 | *** join/#asterisk timahvo1 (~rogue@41.90.148.145) |
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| 07:38.54 | mzb | trung: 1) yes, 2) CID lookup is for callers with CID, of course! |
| 07:39.43 | mzb | the announcement (on occasion also includes big long speech about inconsiderate ppl with-holding their CID, and the numbers they need to dial to un-hide |
| 07:40.15 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
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| 08:25.52 | *** join/#asterisk qakhan (~qakhan@180.178.159.104) |
| 08:26.17 | qakhan | i have TDM800p it doesnt hangup call |
| 08:31.45 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 09:04.15 | ectospasm | qakhan: do you have disconnect supervision on your line? |
| 09:06.02 | kaldemar | ectospasm: it's the other way around |
| 09:07.43 | ectospasm | well, if the telco hasn't enabled disconnect supervision, they'll need busy detection |
| 09:08.34 | ectospasm | ...and hope the telco plays busy/engaged tone after a short period. |
| 09:09.10 | kaldemar | the problem is that his telco does not detect hangups from his side. |
| 09:10.41 | ectospasm | hmmm... how does the telco expect him to hangup? battery drop? polarity reversal? |
| 09:19.12 | qakhan | kaldemar and ectospasm the problem is when i hangup call my card channel hangup but caller phone shows call in progress |
| 09:19.26 | qakhan | until caller hangup his phone my line get busy |
| 09:21.15 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:23.47 | ectospasm | qakhan: do you know how your telco is expecting the hangup signal? Where is this phone line? |
| 09:25.42 | qakhan | ectospasm i m from Pakistan and our Telco is PTCL |
| 09:26.21 | ectospasm | qakhan: do you have opermode=PAKISTAN in /etc/modprobe.d/dahdi.conf? |
| 09:26.35 | qakhan | no |
| 09:26.56 | ectospasm | options wctdm24xxp opermode=PAKISTAN |
| 09:27.33 | ectospasm | I'm not entirely sure PAKISTAN is an available option, check the source |
| 09:28.10 | qakhan | i m usin asterisk 1.4.38 |
| 09:28.30 | ectospasm | Asterisk 1.4 is EOL as of April 21, 2012 |
| 09:28.36 | qakhan | ok |
| 09:29.04 | ectospasm | ...but those options are DAHDI options. |
| 09:29.17 | qakhan | dahdi is 2.4.1 |
| 09:30.09 | ectospasm | that is also old. |
| 09:30.43 | qakhan | i even checked with 2.6.2 |
| 09:30.45 | qakhan | but same |
| 09:31.03 | ectospasm | 2.6.2 does not exist (yet) |
| 09:31.39 | qakhan | sorry 2.6.1 |
| 09:33.10 | ectospasm | yeah, set opermode=PAKISTAN |
| 09:33.18 | ectospasm | ...its in fxo_modes.h |
| 09:34.48 | qakhan | ectospasm what is disconnect supervision alternate? |
| 09:35.43 | ectospasm | disconnect supervision is called many different things throughout the world, it basically is a signal your telco sends when the far end hangs up the call. |
| 09:36.27 | qakhan | if my telco doesnt provide disconnect supervision then? |
| 09:37.27 | ectospasm | you can test whether you have disconnect supervision by connecting either a voltmeter or a POTS handset with a lighted keypad. The voltage will drop to ~0V or the keypad will blink when the far end hangs up. |
| 09:38.07 | ectospasm | but from the description you've given, this isn't a disconnect supervision problem. |
| 09:38.28 | ectospasm | ...it's like you're not providing disconnect supervision to your telco. |
| 09:38.47 | qakhan | ok but if my telco doesnt provide disconnect supervision then what would be solution? |
| 09:39.22 | ectospasm | busydetect and busycount |
| 09:39.32 | ectospasm | ...in chan_dahdi.conf |
| 09:39.36 | qakhan | i used them but same issue |
| 09:39.57 | ectospasm | right, because your telco doesn't hang up the line when you hang up. |
| 09:40.35 | ectospasm | ...they're not getting the hangup signal from you. You'll need to consult with their documentation to determine how you need to signal hangup. |
| 09:41.26 | ectospasm | you may need to adjust the parameters in fxo_modes.h for PAKISTAN, since they look pretty generic (and hence, may not be correct) |
| 09:42.24 | ectospasm | maybe increase the battdebounce parameter |
| 09:42.31 | ectospasm | ...these are just guesses |
| 09:42.55 | ectospasm | Both Pakistan and India tend to have crazy telcos in my experience. |
| 09:45.17 | qakhan | but when i attach phone and call on other phone call get disconnect instantly |
| 09:47.35 | ectospasm | qakhan: right, because the phone is wired to send the right signal. |
| 09:47.51 | ectospasm | the TDM800 is more generic |
| 09:48.22 | ectospasm | you can try increasing the battdebounce parameter, it may be too short for your telco by default |
| 09:50.44 | qakhan | i have # options wctdm24xxp latency=6 only option in /etc/modprobe.d/dahdi.conf |
| 09:51.18 | qakhan | how i increase battdebounce parameter? |
| 09:52.18 | ectospasm | qakhan: check what the current parameter is: cat /sys/module/wctdm24xxp/parameters/battdebounce |
| 09:52.56 | ectospasm | ...then, double it: options wctdm24xxp latency=6 battdebounce=XX |
| 09:53.18 | ectospasm | ...be sure the options line doesn't have a leading '#' |
| 09:53.40 | qakhan | ectospasm is 64 |
| 09:53.45 | qakhan | its 64 |
| 09:53.54 | ectospasm | so double it... 128 |
| 09:54.02 | qakhan | how? |
| 09:54.12 | ectospasm | ...then, double it: options wctdm24xxp latency=6 battdebounce=128 |
| 09:54.41 | qakhan | u mean add battdebounce=128 after options wctdm24xxp latency=6? |
| 09:54.50 | ectospasm | isn't that what it says? |
| 09:55.11 | qakhan | i m confirming from you |
| 09:55.18 | ectospasm | yes. |
| 09:55.20 | *** join/#asterisk gusto (~gusto@88.128.155.22) |
| 09:55.27 | qakhan | i m confuse |
| 09:55.57 | ectospasm | how so? |
| 09:56.03 | qakhan | what? |
| 09:56.10 | ectospasm | why are you confused? |
| 09:56.42 | qakhan | i add this line options wctdm24xxp latency=6 battdebounce=128 in /etc/modprobe.d/dahdi.conf |
| 09:56.44 | qakhan | right? |
| 09:57.26 | ectospasm | Yes, that is correct. |
| 09:57.47 | ectospasm | ...then "service dahdi restart" or whatever is appropriate for your OS. |
| 09:58.06 | qakhan | ok |
| 09:58.22 | qakhan | do u think will it work? |
| 09:58.32 | ectospasm | I haven't a clue, really. |
| 09:58.59 | qakhan | what it exact does? |
| 09:59.02 | ectospasm | but that seems to be the most logical thing to try at this point. If it doesn't work, you'll have to consult with your telco (documentation, support channel, etc.) |
| 09:59.37 | qakhan | they dont help like you |
| 09:59.59 | qakhan | they simply check line by attaching phone |
| 10:00.52 | ectospasm | qakhan: it raises the amount of time (milliseconds) the battery has to drop for the driver to register a disconnect... but may also affect how long the driver will drop battery to register a disconnect |
| 10:01.07 | ectospasm | ...actually, now that I think of it, it may not work. |
| 10:01.16 | ectospasm | ...but it is something to try |
| 10:01.37 | qakhan | y? |
| 10:01.56 | ectospasm | too many questions, just try it and report back. |
| 10:02.26 | qakhan | ok i ll try it when i go to office i ll let you know tomorrow |
| 10:02.31 | qakhan | thanks ectospasm |
| 10:02.41 | ectospasm | good luck!~ |
| 10:08.41 | ectospasm | qakhan: if you get stuck tomorrow, you can always call Digium technical support, they have support hours during your business hours. |
| 10:09.06 | ectospasm | +1.256.428.6161 |
| 10:11.01 | qakhan | hmmm |
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| 12:42.00 | Ice_Strike | Why System() is not executing the file? |
| 12:42.01 | Ice_Strike | System(php /var/lib/asterisk/agi-bin/file.php ${EXTEN}) |
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| 12:58.59 | wdoekes2 | Ice_Strike: php is not executable for the asterisk process or reachable? (e.g. in /usr/local while asterisk has /usr/local not in PATH) |
| 13:00.07 | wdoekes2 | or.. it is executed, but something goes wrong and you don't see the error output.. (try adding >/tmp/error.log 2>&1 ) |
| 13:00.26 | wdoekes2 | or.. you're running asterisk in a chrooted environment, and /bin/sh isn't reachable |
| 13:01.07 | Ice_Strike | wdoekes2 if i type "php /var/lib/asterisk/agi-bin/file.php" on the linux shell it work fine |
| 13:01.14 | Ice_Strike | i also chcked the permissoion |
| 13:01.23 | Ice_Strike | -rwxr-xr-x 1 root root 322 May 13 13:35 file.php |
| 13:02.04 | wdoekes2 | Ice_Strike: so.. you've checked 1 of my 4 possible causes |
| 13:02.19 | Ice_Strike | try adding >/tmp/error.log 2>&1 on where? |
| 13:02.21 | wdoekes2 | the permissions of file.php are irrelevant, btw |
| 13:02.34 | WIMPy | Does your Asterisk run as roor? If not you've found the issue. |
| 13:02.53 | wdoekes2 | WIMPy: why is that? |
| 13:02.56 | Ice_Strike | 14293 root 19 0 50160 31m 6868 S 0.0 4.1 6:07.58 asterisk |
| 13:03.06 | Ice_Strike | its running root |
| 13:03.08 | WIMPy | Ooos. Sorry. Misread. |
| 13:03.22 | WIMPy | Forget that. |
| 13:03.47 | wdoekes2 | Ice_Strike: add those to the System(...) line |
| 13:03.52 | WIMPy | Try to write a log from the script. |
| 13:03.59 | Ice_Strike | will try |
| 13:03.59 | wdoekes2 | e.g. System(echo hi >/tmp/error.log) |
| 13:04.06 | wdoekes2 | that should give you 'hi' in /tmp/error.log |
| 13:04.15 | Ice_Strike | sure |
| 13:04.22 | wdoekes2 | now add the 2>&1 and replace echo hi with php....file.php |
| 13:09.16 | Ice_Strike | hmmm |
| 13:09.20 | Ice_Strike | i've added this |
| 13:09.22 | Ice_Strike | System(echo hi > /tmp/error1.log) |
| 13:09.30 | Ice_Strike | made a call |
| 13:09.36 | Ice_Strike | i dont see file created in /tmp/error1.log |
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| 13:14.40 | wdoekes2 | that would indicate that /bin/sh is not reachable.. are you running in a chroot? |
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| 13:15.07 | wdoekes2 | or that the dialplan is never reached ;) |
| 13:16.58 | Ice_Strike | the dialplan is reached :) |
| 13:17.04 | Ice_Strike | otherwise it wont make a call lol |
| 13:18.01 | Ice_Strike | wdoekes2 Sorry I am kinda new with Linux, how do iu check if I am running in a chroot? |
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| 13:27.57 | Ice_Strike | oh yea i forgot to reload after adding System(echo hi > /tmp/error1.log) |
| 13:28.04 | Ice_Strike | let me try again! |
| 13:29.49 | Ice_Strike | yes i can see error1.log in the /tmp dir |
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| 15:39.05 | wswartz | What's that thing I create to dial a bunch of different people? |
| 15:40.50 | WIMPy | Ampersand |
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| 16:09.23 | verywiseman | I run asterisk 10 , and i have digium card with 24 fxo channels , it is running well ,but there is a issue , pls look at this http://fpaste.org/CcvD/ , i don't know why DAHDI/2-1 is used where DAHDI/1-1 which answered the request? |
| 16:16.57 | WIMPy | Your h extension looks broken. |
| 16:20.24 | verywiseman | WIMPy, i don't have "h" extension |
| 16:21.44 | WIMPy | Or you ignored all warnings about not using a _. extension? |
| 16:26.16 | verywiseman | WIMPy, Thank you very very very much :) |
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| 18:36.07 | Mercury_Zero | Hi. Is there any way to have two separate ALSA channels set up? I have two sound devices, each with their own input and output, which I would like to connect to asterisk. |
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| 19:05.18 | tokozedg | Hello, anyone knows free soft phone capable of opening web page on incoming calls? |
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| 19:16.51 | qakhan | i have TDM800p it doesnt hangup call |
| 19:17.13 | qakhan | it doesnt send disconnect tone |
| 19:17.41 | WIMPy | Disconnect tone? |
| 19:18.02 | qakhan | call disconnect tone |
| 19:18.10 | qakhan | to telco |
| 19:18.15 | WIMPy | What is that? |
| 19:18.27 | qakhan | call disconnect tone or signal |
| 19:18.50 | WIMPy | Are you talking C5 to them or something? |
| 19:19.19 | WIMPy | doubts that any telephone would do that. |
| 19:19.54 | qakhan | ?? |
| 19:20.39 | WIMPy | You do have lines that are suppoed to be used with an ordinary telephone, aren't you? |
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| 19:28.26 | qakhan | yes |
| 19:31.11 | WIMPy | Maybe I'm not that good with the historic stuff, but I haven't heard of phones sending a disconnect tone. |
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| 19:36.51 | qakhan | i told u my issue |
| 19:37.27 | qakhan | call phone shows call in progress and my line get busy |
| 19:38.08 | bn-7bc | correct me if I'm wrong but is there any signaling for call hangup on an pstn line, is is the line just considered as hung up a few secs after the circuit from switch to subscriber is broken? |
| 19:39.05 | WIMPy | It's only about impedance/current. |
| 19:39.14 | tokozedg | qakhan: try to change cptone profiles if it has any. I had the same problem, and Taiwan profile matched with my country. Or at least you can measure disconnect signal and set to it. |
| 19:43.46 | qakhan | tokozedg what is cptone? and how i can measure disconnect signal and set it |
| 19:55.42 | qakhan | tokozedg U there? |
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| 20:56.41 | darrenlooby | Hello all. I'm running "sip show peers" and the first column - Name/username is too short... is there a way I can change the length, as we have quite long usernames to deal with |
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| 21:01.13 | juliocesarlhg | hi |
| 21:01.26 | juliocesarlhg | which is the best way to call to germany? |
| 21:01.44 | juliocesarlhg | land lines, is there a freeservice like gv? |
| 21:15.18 | WIMPy | Some ITSP like e.g. sipgate. Their EU flat is 8.90. |
| 21:28.25 | juliocesarlhg | gv is free forever? |
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| 21:37.19 | [TK]D-Fender | [17:28]juliocesarlhggv is free forever? <- no. |
| 21:37.45 | juliocesarlhg | ok |
| 21:37.47 | juliocesarlhg | thanks |
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| 22:47.45 | darrenlooby | Hello all. I'm running "sip show peers" and the first column - Name/username is too short... is there a way I can change the length, as we have quite long usernames to deal with |
| 22:48.12 | WIMPy | May the source be with you. |
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| 23:11.22 | Ice_Strike | What the transfer speed in Mb/sec for 10 concurrent calls? |
| 23:11.29 | Ice_Strike | in 1 hour |
| 23:11.36 | Ice_Strike | or 1 minue |
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| 23:12.53 | WIMPy | That depends on connection type, codec and packet size. Try to ask google for a voip bandwidth calculator. |
| 23:13.06 | Ice_Strike | Oh |
| 23:13.07 | Ice_Strike | Thanks |
| 23:14.59 | Ice_Strike | How do I find out what the packet size? |
| 23:15.24 | WIMPy | You configure it. |
| 23:15.34 | WIMPy | The default is usually 20ms. |
| 23:16.12 | WIMPy | I always set it down to 10ms to minimise the chance for annoying delays. |
| 23:16.15 | Ice_Strike | 1.63 Mbps for 10 concurrents call |
| 23:16.16 | Ice_Strike | not bad |
| 23:16.27 | Ice_Strike | According to http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 23:16.54 | Ice_Strike | is VPS safe to use? |
| 23:17.03 | Ice_Strike | or shall i get decicated server |
| 23:17.11 | prgmrchris | Ice_Strike: not recommended for production |
| 23:17.31 | Ice_Strike | well not for production - just temporary for 1 month |
| 23:17.34 | jpsharp | You can get away with VPS with a very low call volume. |
| 23:18.00 | Ice_Strike | will be making 10 concurrent calls for 24 hours for 1 month though |
| 23:18.05 | WIMPy | There have been quite some issues with timing in the past, but that may have been sorted. |
| 23:18.41 | Ice_Strike | including recording |
| 23:18.43 | jpsharp | It mostly depends on how loaded your VPS host server is and which virtualization platform they use. |
| 23:18.44 | WIMPy | I haven't heard of these issues lately. |
| 23:19.00 | Ice_Strike | jpsharp it will be XEN |
| 23:19.47 | jpsharp | I've run 8 concurrent calls through my VMWare Asterisk system here, but my VMWare host barely breaks a sweat. |
| 23:19.48 | Ice_Strike | which vps do u recommend |
| 23:21.05 | Ice_Strike | 1.63 Mbps for 10 concurrents call - how much bandwidth is that for 1 month |
| 23:21.39 | prgmrchris | do the math |
| 23:22.18 | Ice_Strike | 70416 if im correct |
| 23:22.34 | WIMPy | That's a relly good porn library. |
| 23:22.41 | Ice_Strike | lmao lol |
| 23:24.37 | Ice_Strike | which voip provider you guys with? |
| 23:25.33 | Ice_Strike | flowroute is quite good |
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| 23:59.41 | _370z-hsv | dead as hell in here |
| 23:59.42 | _370z-hsv | lulz |