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| 02:38.15 | trulsk | is it frowned upon to ask 'who can code keep-place-in-queue-called-back-automatically for me' here? are there other forums that are more suited? I saw the bounty-system, but that looked more for feature-requesting etc |
| 02:40.53 | *** part/#asterisk Buuyo (~electrum@1c.posi.xxx) |
| 02:43.15 | jsmith | trulsk: The asterisk-biz mailing list is probably the most appropriate place to ask that |
| 02:44.56 | trulsk | cool. subscribed, thanks! |
| 02:47.15 | trulsk | and I should probably upgrade from 1.2.10. Are there any knows security issues with old releases or am I just missing out on sweet stuff? |
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| 02:50.27 | trulsk | ive skimmed through changelog, and saw nothing but issues with sample configuration and http. |
| 02:50.40 | trulsk | changelog from 1.2.10 till now is loooong tho |
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| 05:55.21 | dandate2 | is it preferable to have a hardware timer for iax2 trunking |
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| 06:26.47 | ChannelZ | OT: Any recommendations on unmanaged switches with PoE for at least 12 ports? |
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| 06:37.12 | dandate2 | catalyst 3550 is pretty cheap |
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| 06:41.02 | bulkorok | hi |
| 06:44.46 | ChannelZ | dandate2: thanks, looking at it now |
| 06:50.43 | dandate2 | make sure your catalyst says POE on the front, have the ebay seller send you a pic |
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| 07:04.27 | ChannelZ | It's weird, D-Link shows two smart switches on their website but they've been discontinued for years. |
| 07:07.09 | jacc0 | we use netgear |
| 07:07.18 | jacc0 | for poe |
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| 07:48.41 | ChannelZ | Yeah that's where I'm leaning, they seem to have bang for the buck |
| 07:50.35 | ChannelZ | I've been using a couple of their unmanaged dumb desktop switches (16 port/8-port PoE) for years, I was building a company on the cheap |
| 07:52.06 | ChannelZ | Second power adapter has failed so I'm just looking to replace them all with something more solid |
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| 08:33.21 | schmidts | good morning |
| 08:33.39 | ChannelZ | ahoyhoy |
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| 08:54.29 | dandate2 | is it preferable to have a hardware timer for iax2 trunking |
| 09:02.20 | schmidts | dandate2 yes |
| 09:02.45 | jacc0 | we don't use hardware timing but we do use dahdi |
| 09:02.59 | dandate2 | why is it better than ztdummy ? |
| 09:04.21 | schmidts | dandate2 i dont really know if you need hardware timing but atleast i had very big timing problems with iax at all |
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| 09:19.26 | *** join/#asterisk acecile (~acecile@adsl02.metz.linbox.com) |
| 09:19.33 | acecile | hi guys |
| 09:20.19 | acecile | I'm having an issue with asterisk |
| 09:20.25 | acecile | and call forward |
| 09:20.50 | acecile | when calling from one extension to another extension with cf set to a mobile phone |
| 09:20.53 | acecile | everything works |
| 09:21.27 | acecile | when calling a DID used by an extension with CF set to my mobile |
| 09:21.35 | acecile | the mobile set, I can pick up the call |
| 09:21.37 | acecile | no sound |
| 09:21.48 | acecile | here is the astersik debug 10 trace |
| 09:21.50 | acecile | http://paste.debian.net/172143/ |
| 09:22.03 | acecile | any help would be appreciated guys... I'm no asterisk guru |
| 09:28.28 | kaldemar | you left out the interesting parts. enable sip debug and make a new call, and pastebin the whole call. |
| 09:30.15 | acecile | got it working |
| 09:30.18 | acecile | nat issues.... |
| 09:31.18 | Digweed | Does anyone know a nice script that visualizes the total of active calls on my Asterisk servers? |
| 09:31.25 | Digweed | Would like to display that on a screen at the office. |
| 09:31.37 | acecile | progressinband=yes fixed the issue |
| 09:32.04 | acecile | does it makes sense to you ? |
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| 09:35.37 | din3sh | morning all |
| 09:38.31 | kaldemar | acecile: you may have masked the real issue with progressinband. |
| 09:39.03 | acecile | ah |
| 09:39.11 | acecile | which is a nat issue ? |
| 09:40.56 | kaldemar | i don't know anything about your setup. so no idea. |
| 09:41.14 | acecile | ippi siptrunk provider |
| 09:41.23 | acecile | one box, two lan |
| 09:41.27 | acecile | a separate lan for phones |
| 09:41.37 | acecile | the other lan is connected to our main lan |
| 09:42.03 | acecile | so it goes through a router to reach the sip trunk provider |
| 09:42.25 | kaldemar | most likely related to NAT. |
| 09:43.33 | acecile | so it would be better to use publicip + localaddress |
| 09:43.38 | acecile | than progressinband |
| 09:43.39 | acecile | ? |
| 09:44.43 | kaldemar | more like you have to configure your box properly. |
| 09:44.48 | kaldemar | ~sipnat |
| 09:44.48 | 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. |
| 09:45.18 | acecile | thank you sir :) |
| 09:46.41 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:53.51 | acecile | the first tutoriel worked |
| 09:54.02 | acecile | I'm "progressinband" free :p |
| 09:59.30 | Digweed | In case someone missed my question, I would like to current active calls on a screen at the office. We have 16 active Asterisk servers so it would be nice to display the current of each and a total. Is there software available for this? |
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| 10:11.30 | acecile | do you know any good french language pack ? |
| 10:11.43 | acecile | even a not free one |
| 10:11.50 | acecile | I don't like much the ones included |
| 10:13.47 | Digweed | Hmm, when I setup a call using my softphone to a number that does not exist, Asterisk receives 404 from the provider. The softphone however receives 503. Why is that? Asterisk 1.8 |
| 10:15.26 | Digweed | Seems like Asterisk translates the 404 to 503 for my softphone. |
| 10:16.56 | schmidts | Digweed there was something on the asterisk-dev mailinglist depending to this problem |
| 10:17.54 | schmidts | Digweed and about your concurrenct calls, take a look at MRTG or CACTI and using the ami interface for it, so you can also graph this values |
| 10:18.36 | Digweed | Ok schmidts sounds good, I will definitely check that. Regarding the cause codes, I guess I'll Google if you're saying it has been discussed on the mailinglist. Should be indexed on the web. |
| 10:19.20 | schmidts | digweed it was about last week, so i dont know ifs allready on the google store ;) |
| 10:20.00 | Digweed | Ok, if it's not on Google can I find it somewhere hah, not familiar with the mailinglist... |
| 10:21.18 | Digweed | http://lists.digium.com/pipermail/asterisk-dev/2012-May/subject.html |
| 10:22.57 | Digweed | Yep here we go http://lists.digium.com/pipermail/asterisk-dev/2012-May/055309.html |
| 10:25.53 | schmidts | thats what i meant ;) |
| 10:26.13 | Digweed | You're extremely helpful man, really appreciate it. |
| 10:32.45 | schmidts | your welcome ;) |
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| 11:29.36 | Digweed | schmidts, once again thank you. i read the mailinglist thread, that pointed to https://issues.asterisk.org/jira/browse/ASTERISK-19914 (created because of the mailinglist), that has a patch, i applied that patch |
| 11:29.37 | Digweed | resolved |
| 11:29.40 | Digweed | thank you so much |
| 11:30.23 | WIMPy | I guess that's a topic that will haunt us for a longer time in other places as well. |
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| 11:32.18 | Digweed | Indeed WIMPy ... |
| 11:33.04 | Digweed | Can't imagine how happy I was to see "Status-Code: 404" in Wireshark hahahahaha. |
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| 11:54.02 | mjordan | trulsk: all security issues are reported on asterisk.org/security. And yes, there are probably lots of known security issues with 1.2. |
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| 12:20.33 | rgsteele | Would it be a bug if calls are being answered but the disposition is always set to "NO ANSWER" and billsec is always set to 0? The start time of the call is noted, and the duration increases normally as the call progresses, but due to the faulty disposition, the 'answer' and 'end' CDR variables are never set. |
| 12:24.31 | rgsteele | This is asterisk 1.6.2.5, with some newer patches applied (here is the changelog: http://changelogs.ubuntu.com/changelogs/pool/universe/a/asterisk/asterisk_1.6.2.5-0ubuntu1.4/changelog) |
| 12:26.08 | [TK]D-Fender | ancient release of a dead branch |
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| 12:33.34 | singler | rgsteele: there was some sort of bug, you'll need to patch manually or upgrade |
| 12:35.26 | rgsteele | singler: Are you referring to https://issues.asterisk.org/view.php?id=13665 ? In my case, billsec is always 0 (although that's not abnormal - in the older asterisk system on which everything works, billsec is always 0 even though the disposition is correctly set as "ANSWERED") |
| 12:35.38 | rgsteele | And, that apparently applies to 1.4 only |
| 12:36.04 | singler | I do not know which bug, I remember I had similar problem, and it was fixed later |
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| 12:43.47 | din3sh | how can i disable transfer to another context? |
| 12:44.53 | WIMPy | din3sh: Err, what? |
| 12:48.13 | pithagorians | <WIMPy> hi |
| 12:49.19 | pithagorians | <WIMPy> still no solution to my issue. wrote on forum..maybe somebody knows there .. |
| 12:49.45 | WIMPy | Which one was that? |
| 12:49.54 | pithagorians | http://forums.asterisk.org/viewtopic.php?f=1&t=82906 |
| 12:50.51 | WIMPy | Right. |
| 12:51.17 | WIMPy | Did you answer my question if your local caller ID is part of the external one? |
| 12:51.59 | pithagorians | it isn't |
| 12:52.16 | WIMPy | Then you have to do it manually. |
| 12:52.25 | pithagorians | local caller id is going to be Person Name <number> |
| 12:52.39 | pithagorians | like John Doe <700> |
| 12:52.39 | WIMPy | If it's all SIP devices you could use setval in sip.conf to set the external CID. |
| 12:52.39 | pithagorians | for ex |
| 12:56.23 | pithagorians | setvar=EXTERNAL_CALLERID="User One" <3012323434> |
| 12:56.25 | pithagorians | like this? |
| 12:56.45 | WIMPy | For example. |
| 12:57.03 | WIMPy | Then you can use that variable in your dialplan where you call out. |
| 12:57.13 | pithagorians | understand |
| 12:57.15 | pithagorians | thx man ! |
| 12:57.17 | pithagorians | :) |
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| 13:22.02 | rgsteele | singler: Hm... upgrading from 1.6.2.5 to 1.6.2.9 didn't fix the issue. :-/ |
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| 13:24.56 | singler | rgsteele: and why you do not upgrade to newest 1.6.2.x? or even to 1.8.x or 10.x? |
| 13:25.43 | [TK]D-Fender | rgsteele, <------ latest in branch |
| 13:25.45 | rgsteele | singler: There's a webapp that's been internally written that doesn't work with the 1.8/1.10 branches. Much as I'd like to, I have to wait on some developer time for that move. |
| 13:25.52 | [TK]D-Fender | rgsteele, You aren't even looking into DOUBLE DIGIT |
| 13:26.25 | rgsteele | I guess I'll try and backport 1.6.2.22 or .24 or whatever's the latest in that branch. |
| 13:27.07 | [TK]D-Fender | 1.6.2.24 |
| 13:27.13 | [TK]D-Fender | dang numlock |
| 13:28.16 | wdoekes2 | if you're going to run 1.6.2 latest, you may need to undo r319652 |
| 13:29.20 | singler | what does it do? |
| 13:29.28 | rgsteele | wdoekes2: Will keep that in mind. |
| 13:29.30 | wdoekes2 | see this https://issues.asterisk.org/jira/browse/ASTERISK-18136 |
| 13:29.38 | *** join/#asterisk topro (~quassel@host-62-245-142-50.customer.m-online.net) |
| 13:30.10 | rgsteele | The only difference between this box and the one that works is this one's using dahdi, the older one's using zaptel. |
| 13:30.31 | rgsteele | Dunno if that's relevant or not, though, since calls themselves actually work, it's just the CDR variables that aren't being set right. |
| 13:35.12 | file | oh god CDRs |
| 13:39.17 | rgsteele | Hmmm |
| 13:39.24 | rgsteele | same result with 1.6.2.24 |
| 13:39.31 | rgsteele | disposition=NO ANSWER :-/ |
| 13:39.42 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:41.50 | rgsteele | What's weirder is when the call ends, it inserts the CDR into the DB with disposition set to ANSWERED..... |
| 13:44.01 | rgsteele | sighs |
| 13:46.14 | rgsteele | Hm |
| 13:46.18 | rgsteele | It's only for inbound calls |
| 13:46.44 | rgsteele | Outbound calls show all the correct info |
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| 13:50.17 | gtTuna | will I need to make any drastic changes to my dialplan code when upgrading from 1.6.2 to 1.8? |
| 13:50.44 | rgsteele | gtTuna: Test it :) |
| 13:50.59 | gtTuna | going to when I get the chance this weekend hopefully |
| 13:51.22 | gtTuna | need to make a backup clone of one of my older servers anyways |
| 13:52.17 | gtTuna | reading through all of the changelog, i dont immeidately see anything that will cause huge issues -- just wanted to know if anyone had any pointers/tips/caveats to watch out for |
| 13:53.47 | Gugge | i have "exec_after_rotate=gzip -9 ${filename}.2" in logger.conf, and after upgrade from 1.6.2.something to 1.8.12.0 the logs are not compressed anymore, what am i missing? |
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| 14:15.44 | timholum | Hello, I have exten => s,n,Dial(SIP/101200&SIP/101208&SIP/101209&SIP/101223&SIP/101201&SIP/101203&SIP/101204&SIP/101202,15,tTM(ctl-answer-mac)), In my ctl-answer-mac I have an agi script, But that agi script is unable to access variables that I set in the main context, is there a way to pass that through? |
| 14:18.46 | timholum | here is a pastebin of the relivent dialplan http://pastebin.com/edtNsF4Z |
| 14:18.50 | rgsteele | Hm, I do see that it works for inbound calls as well on the old Asterisk installation. The output of 'core show channel <channelname>' on both shows the following differences: http://pastie.org/4001923 |
| 14:21.17 | rgsteele | I don't really see much in there that identifies a culprit, and the configs are identical except for the newer installation using DAHDI instead of Zaptel. |
| 14:25.32 | rgsteele | I'm guessing that the difference in value for 'Application' and 'Data' are relevant, but I'm not quite sure how |
| 14:26.28 | kaldemar | timholum: it's not a matter of context, but channel. variables are assiciated with a channel unless declared with global or shared functions. also, the macro is executed on the called channel, so any variables you set for the calling channel won't show up in the macro or whatever it executes. |
| 14:26.37 | kaldemar | timholum: use arguments for the macro. |
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| 14:27.22 | timholum | kaldemar: thanks, |
| 14:28.18 | rgsteele | What really gets me though is that when the call ends, the value of 'disposition' is 'ANSWERED' when it's inserted into the CDR table in the database. |
| 14:29.04 | rgsteele | Maybe it's just a display bug? |
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| 14:29.58 | *** join/#asterisk phanohanover (phanohanov@modemcable096.110-80-70.mc.videotron.ca) |
| 14:30.33 | phanohanover | Freepbx on asterisk can't fix this: manager.c:2423 authenticate: 127.0.0.1 failed to authenticate as 'user' |
| 14:30.36 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:30.39 | phanohanover | Someone knows how6 |
| 14:31.12 | [TK]D-Fender | phanohanover, Clearly the user or pass doesn't match. Fix it |
| 14:31.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:31.19 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:31.47 | [TK]D-Fender | \o/ |
| 14:31.56 | [TK]D-Fender | NEXT!@!! (c) BKW |
| 14:32.04 | phanohanover | match between amportal and manager.conf are the same...but in manager.conf, I have both admin and user setup as well as manager.conf exactly the same... |
| 14:32.31 | [TK]D-Fender | ~freepbx |
| 14:32.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:32.39 | phanohanover | sorry as well as in amportal API php page |
| 14:33.12 | phanohanover | ok, I get it...thanks guys |
| 14:34.01 | rgsteele | It also looks like /var/log/asterisk/cdr-custom/Master.csv inserts 'ANSWERED' into the logs... so perhaps it is just a display bug. |
| 14:36.52 | rgsteele | Wonder if murf ever lurks in here - he's Mr. CDR. |
| 14:40.25 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 14:40.49 | rgsteele | Oof... this isn't good... asterisk gets itself into an infinite loops somehow and pegs the cpu on all 3 1.6 versions I've tested: http://pastie.org/4002063 |
| 14:41.36 | rgsteele | looks like a dahdi bug |
| 14:45.57 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 14:48.24 | wdoekes2 | and you're not using SIP_HEADER or SIPCHANINFO ? what happens when you increase verbosity? infloop in the dialplan? |
| 14:48.42 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-32-16.user.veloxzone.com.br) |
| 14:49.25 | *** join/#asterisk darkski3z (~mhb@fsf/member/darkskiez) |
| 14:49.26 | *** join/#asterisk SunTsu_ (miyamoto@unaffiliated/suntsu) |
| 14:51.33 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 14:57.10 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:57.11 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:57.22 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
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| 15:05.10 | rgsteele | wdoekes2: I am using SIP_HEADER |
| 15:05.31 | rgsteele | s,1,Set(TOLLFREENUM=${CUT(SIP_HEADER(TO):5,@,1)}) |
| 15:08.20 | rgsteele | Could it possibly be a phantom ring? |
| 15:08.50 | *** join/#asterisk timahvo1 (~rogue@41.80.62.248) |
| 15:09.36 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:10.31 | *** join/#asterisk din3sh (din3sh@41.136.84.184) |
| 15:12.38 | din3sh | WIMPy: asterisk.a is connected to pstn, call from asterisk.b comes throught to asterisk.a via sip trunk, i dont want the users in asterisk.a to be able to transfer the call to pstn. i want the call to remain local to asterisk.a |
| 15:13.26 | fubada | hey guys, my * 10.4 is segfaulting when using confbridge (and possibly during other sip calls), heres the dmesg output: http://pastie.org/4002244 |
| 15:13.36 | fubada | happens every time i confbridge about 60 seconds into it |
| 15:13.46 | fubada | asterisks all of a sudden dies and restarts |
| 15:14.10 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 15:14.42 | wdoekes2 | fubada: http://lists.digium.com/pipermail/asterisk-dev/2012-May/055387.html ? |
| 15:15.00 | fubada | nice |
| 15:15.03 | fubada | niiiiice |
| 15:15.06 | fubada | you da man |
| 15:15.18 | WIMPy | din3sh: Then don't allow the users on *b to dial *a. |
| 15:15.39 | WIMPy | Or do you have Features activated on *a that kick in? |
| 15:15.59 | din3sh | they are on remote locations |
| 15:16.18 | WIMPy | ? |
| 15:16.44 | din3sh | say the 2 asterisks are in different countries |
| 15:16.47 | WIMPy | We require the full picture. |
| 15:17.05 | din3sh | on a vpn |
| 15:17.46 | din3sh | they would be able to call each other, but i cannot terminate the calls on the respective pstns the asterisks are connected |
| 15:18.15 | din3sh | or i dont want the calls to be terminated on pstn |
| 15:18.23 | din3sh | to put it that way |
| 15:18.44 | WIMPy | Ok, you've got two boxes where you can dissalow that. |
| 15:19.03 | din3sh | by not using tT options? |
| 15:19.29 | WIMPy | Do you have a reason to use them? |
| 15:19.42 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 15:19.47 | din3sh | not really |
| 15:20.10 | din3sh | i have tried setting __Transfer_context |
| 15:20.23 | WIMPy | And I'm pretty certain, you don;t want to use both, anyway. |
| 15:20.30 | din3sh | to another local context having no route to pstn |
| 15:20.38 | edge | I setup a call queue and when a DAHDI call comes in and is put into the queue the sets lose the Call ID information, and instead say "Asterisk - Asterisk" , how do i pass that information along to the queue? |
| 15:21.46 | [TK]D-Fender | ediIt doesn't lose anything. |
| 15:21.51 | [TK]D-Fender | edge, It doesn't lose anything. |
| 15:22.01 | [TK]D-Fender | edge, It never had it |
| 15:22.30 | edge | [TK]D-Fender: Guess i should rephrase, how can i ask DAHDI to pass that information to the queue application |
| 15:22.47 | din3sh | if i don't Tt option, the phone isnt not able to transfer with transfer key set in features.conf, but the phone can still transfer via its own transfer button (sip refer?) |
| 15:23.05 | *** join/#asterisk danfromuk (~IceChat77@2.27.7.133) |
| 15:23.07 | [TK]D-Fender | edge, "usecallerid=yes", "callerid=asreceived" |
| 15:23.23 | WIMPy | din3sh: correct |
| 15:23.40 | edge | [TK]D-Fender: does that go into chan_dahdi? |
| 15:23.46 | [TK]D-Fender | yes |
| 15:24.02 | danfromuk | Hi, I know this isnt really the place, but I urgently need a good quality sip trunk as Magrathea is currently down. Can anyone recommend a good uk based provider that can set up a sip trunk in a few minutes? |
| 15:24.23 | din3sh | WIMPy: there's no way to block that call from being transfered then? |
| 15:25.13 | WIMPy | danfromuk: sipgate? |
| 15:25.22 | Qwell | danfromuk: They aren't down, just out building planets. |
| 15:25.41 | WIMPy | din3sh: sip.conf, "allowtransfer" IIRC. |
| 15:25.42 | edge | Thank you very much [TK]D-Fender |
| 15:27.42 | din3sh | WIMPy: thanks will try that tomorrow at office |
| 15:30.36 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:39.03 | *** join/#asterisk Micc (~Mic@c-24-19-33-189.hsd1.wa.comcast.net) |
| 15:39.28 | Micc | I've got a problem with some Grandstream gx2100. |
| 15:40.23 | Micc | After a random amount of time, sometimes 1, sometimes 3 or 4 minutes into the call, it turns into star wars sounding. The codec just goes weird, and it doesn't matter which codec we use, I've tried g722 and g711. |
| 15:40.44 | Micc | I've played with the voice frames per tx and it doesn't seem to change the results at all. |
| 15:41.18 | Micc | Asterisk is 1.8.9.2. |
| 15:41.33 | Micc | And the phones work fine on ring central. |
| 15:42.01 | Micc | And we have about 500 other sip devices on the same server working fine for over a year. |
| 15:42.31 | Micc | I know grandstream are the bottom of the barrell, but I always thought they at least worked. |
| 15:43.17 | Micc | We've never done service for a grandstream gxp2100, just the handy tones. |
| 15:43.36 | Micc | So I don't have anything else to compare it to. |
| 15:44.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:44.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:45.13 | *** join/#asterisk bluregard (~matt@c-71-201-99-216.hsd1.il.comcast.net) |
| 15:45.32 | bluregard | hello all |
| 15:52.46 | bluregard | If I blind transfer a call to an extension, is there any way to get at the user-agent for the device that performed the transfer, similar to the way ${BLINDTRANSFER} gives you the channel of the device that did the transfer? |
| 15:53.12 | leifmadsen | bluregard: set the value to a channel variable that is inherited I guess |
| 15:53.14 | bluregard | I thought ${SIPCHANINFO(USERAGENT)} would do it, but it gives me the UA of the device that was transfered which does make sense. |
| 15:53.22 | leifmadsen | bluregard: see above |
| 15:53.30 | anonymouz666 | I wonder why CentOS 6 with asterisk is about 50% faster than CentOS 5 |
| 15:53.43 | leifmadsen | Set(_thisUserAgent=${SIPCHANINFO(useragent)}) |
| 15:54.17 | bluregard | yeah |
| 15:55.23 | leifmadsen | Set(_${CHANNEL(name)_UserAgent=${SIPCHANINFO(useragent)}) |
| 15:55.48 | leifmadsen | perhaps do that, then you can: ${${BLINDTRANSFER}-UserAgent} or something |
| 15:55.57 | bluregard | hey yeah that's a good idea. I'm already grabbing the channel info. |
| 15:55.58 | bluregard | right |
| 15:56.31 | bluregard | that should work beautifully, thank you. |
| 16:00.23 | Micc | Anyone have any ideas on my star wars audio on the grandstream? |
| 16:01.51 | WIMPy | Did you pay royalties to Lukas? |
| 16:04.08 | Micc | Haha, no, I bet he'll be mad about that. |
| 16:04.23 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v023-129.mobile.uci.edu) |
| 16:04.38 | Micc | It sounds just like a light saber battle or tie fighter battle. |
| 16:06.56 | Micc | after ten years doing this, I thought I had heard everything. But I can't figure out how this is happening. I don't think its packet loss. Its just so strange. seems like its just a bad phone, but they say it works fine with ring central. |
| 16:07.07 | Micc | Does anyone know what server platform ring central uses? |
| 16:09.24 | *** join/#asterisk lando (~lando@c-98-254-147-186.hsd1.fl.comcast.net) |
| 16:12.03 | Micc | I suppose I should try a different server, maybe an asterisk 10 server. |
| 16:13.07 | Qwell | or a different phone :) |
| 16:14.23 | [TK]D-Fender | ^^^^ |
| 16:14.36 | [TK]D-Fender | remove the first piece of the equation and validate |
| 16:17.34 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:23.47 | *** join/#asterisk dandate2 (~dan@180.190.172.203) |
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| 16:24.10 | dandate2 | if theres anyone from asterisk.org here, http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ is a broken link |
| 16:29.00 | leifmadsen | dandate2: where did you find that? |
| 16:29.10 | leifmadsen | because it is supported to be dahdi/linux-complete |
| 16:29.43 | dandate2 | im not able to download dahdi linux from http://www.asterisk.org/downloads |
| 16:29.51 | dandate2 | link is broken |
| 16:30.39 | sruffell | whoops…my bad. |
| 16:30.54 | sruffell | I'll fix it, thanks dandate2 |
| 16:32.32 | sruffell | hmm... |
| 16:33.12 | sruffell | actually…it might just be because the server is down for maintenance right now. |
| 16:33.12 | sruffell | that would be the *second* time I've gotten hit by that. |
| 16:33.12 | sruffell | today |
| 16:34.16 | sruffell | http://lists.digium.com/pipermail/asterisk-users/2012-May/272548.html |
| 16:34.20 | Qwell | sruffell: heh, you and me both |
| 16:35.29 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:35.29 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 16:37.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:43.56 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:45.01 | bluregard | liefmadsen: so that does work as intended for the most part. The problem is where to set the initial var to make sure its the correct UA. |
| 16:45.22 | leifmadsen | bluregard: that's a logic problem |
| 16:45.49 | leifmadsen | if you're making it unique for each channel you shouldn't have any issues |
| 16:47.28 | bluregard | it is unique since the first portion of the var is the chan name. But if I set the var prior to the dial it ends up being the UA of the calling device, setting it after the blind xfer results in the same problem as before where its the UA of the transferred device. |
| 16:47.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:47.44 | leifmadsen | right |
| 16:47.57 | bluregard | I don't have anywhere else in the dialplan to set the var |
| 16:49.53 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 16:50.57 | bluregard | the idea is to be able to get the user-agent of the device that initiated a blind transfer to the parking ext in order to use the correct agi to push a notification to the phone with the ext the call was parked on. |
| 16:52.02 | Qwell | bluregard: skip all that and just buy Digium phones, and use the builtin parking application. :p |
| 16:53.15 | bluregard | qwell: yes that would be where the idea came from. |
| 16:54.37 | bluregard | I have everything working great with both Polycom and Cisco SPA phones, but they both use completely different methods to push notifications to the phones so I have to have some way of telling the agi what type of phone did the park. |
| 16:57.12 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-pqnbkvoyqmrrrvnx) |
| 17:00.49 | bluregard | I suppose if all else fails I could send cisco phones to one parking ext and polycoms to another |
| 17:09.57 | talntid | sucks having to do things like that, though |
| 17:10.53 | bluregard | yeah, I just tried it. Its not by any means an elegant solution but it does work. |
| 17:12.01 | bluregard | I can't think of any way to get at the correct user-agent after a blind xfer though. |
| 17:14.42 | *** join/#asterisk jtel (4727ec3e@gateway/web/freenode/ip.71.39.236.62) |
| 17:15.01 | jtel | has any one set up realtime for parking |
| 17:15.09 | talntid | heh |
| 17:15.13 | talntid | seems to be the topic today. |
| 17:15.50 | bluregard | naa, mine isn't realtime |
| 17:20.17 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:21.04 | rgsteele | Hm, maybe the fact that the Asterisk CLI shows the disposition as "NOT ANSWERED" is more than just a display issue. My call monitor shows the call in a state of 'bridging' instead of 'talking'. |
| 17:25.12 | *** join/#asterisk wonderworld (~ww@dsdf-4db530e4.pool.mediaWays.net) |
| 17:25.13 | justdave | if I'm using DUNDi to look up what server can handle an extension via a switch=> statement, and I get two weighted responses for an extension number, and the lower-numbered weight one times out, it automatically falls through to the next higher numbered weight... which is good... |
| 17:25.26 | justdave | but is there a way to control how long it takes to time out before falling through? |
| 17:26.01 | justdave | seems like it tries for about 30 seconds, and a user isn't going to wait that long. I've got a good network, if it hasn't answered in 2 or 3 seconds it's not going to. :) |
| 17:28.22 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:29.00 | jtel | with asterisk parking can you do more then one context |
| 17:29.42 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:29.42 | bluregard | I got it. I'm getting the IP address of the peer that parked the call with some cutting of ${BLINDTRANSFER} and the ${SIPPEER} func which can also give you the UA. |
| 17:32.26 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:32.51 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:33.36 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.92.36.dsl.dyn.forthnet.gr) |
| 17:33.50 | *** join/#asterisk din3sh (din3sh@41.136.83.24) |
| 17:36.39 | bluregard | I can't believe I overlooked that |
| 17:40.57 | justdave | looks like I can set ${DUNDIDIALARGS} but the only timeout value I see that I can pass to Dial() appears to affect ringring as well as connecting to next hop... |
| 17:41.17 | justdave | wouldn't be good to hang up on a ringring phone after 5 seconds :( |
| 17:41.48 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:42.41 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 17:43.01 | slav3_kitten | Hi, i have a couple of stupid questions before i jump into pbx stuff feet first |
| 17:43.19 | bluregard | shoot |
| 17:44.39 | slav3_kitten | Can asterisk do something like customer calls in on a 20 channel voip did number, dials an extension, that extension dials out to agent's cell phone, if it's not answered in x seconds sends the customer call to voice mail box. and if the agent calls back on the office did number they get their voice mail and the ability to dial internal pbx extensions? |
| 17:45.12 | bluregard | sure |
| 17:45.19 | slav3_kitten | also can asterisk be setup to give different virtual agent greetings based on time of day/holiday/etc? |
| 17:45.34 | bluregard | yep |
| 17:46.12 | slav3_kitten | any books you would recommend for learning the burnt of asterisk that i can get in pdf format? |
| 17:46.16 | bluregard | the best thing you can do is get yourself a copy of the book |
| 17:46.21 | bluregard | ~book |
| 17:46.21 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:46.40 | bluregard | then set up an asterisk box and start playing with it |
| 17:46.57 | slav3_kitten | oh and one final question. asterisk vs freeswitch |
| 17:47.04 | bluregard | well gee |
| 17:47.33 | bluregard | considering you're in #asterisk the answer should be obvious |
| 17:47.43 | slav3_kitten | reading the freeswitch stuff leads me to believe asterisk is a tad unstable... but i think that might be only in situations involving way larger operations than i will ever do |
| 17:48.17 | bluregard | I've deployed asterisk with hundreds of extensions and concurrent calls and never had a stability issue |
| 17:48.48 | slav3_kitten | answers that question :) |
| 17:48.49 | bluregard | I usually stay away from bleeding edge in production though. |
| 17:49.10 | slav3_kitten | sorry, i'm entirely new to voip and software pbx's |
| 17:49.37 | bluregard | they're alot of fun. In my opinion at least. |
| 17:49.45 | Qwell | slav3_kitten: Also, about that document... Competing products have never been known to lie. :) |
| 17:50.12 | slav3_kitten | ideally i want to run a 6 cisco phone voip network in the house connected to a single pots line via my 2621xm with a vic-2fxo card in it |
| 17:50.21 | Qwell | yeah, ditch the Ciscos |
| 17:50.28 | bluregard | yeah, nuts to those. |
| 17:50.37 | slav3_kitten | but i have a friend with a gaming business who wants voip service to handle a bunch of shit |
| 17:50.38 | bluregard | I have a spa509g here that sucks |
| 17:50.51 | slav3_kitten | cisco isn't going to be ditched. they are all sip compatible |
| 17:50.59 | bluregard | but they suck |
| 17:51.17 | slav3_kitten | but i have them, my entire house network is cisco gear + mikrotik for wireless |
| 17:51.23 | bluregard | I love polycoms and I've heard a lot of good things about digium phones, but never used one. |
| 17:51.40 | slav3_kitten | it's not going to change in any way shape or form as i have them already and i have limited to no income most the time |
| 17:51.48 | bluregard | so be it |
| 17:52.13 | bluregard | I'm opposed to paying twice as much for green hardware but if you already have it then... |
| 17:52.28 | slav3_kitten | i am way open to other phones for anything i do for anyone else because i've read a shit ton of issues with quirky shit |
| 17:53.17 | slav3_kitten | i mean i got my 2621xm maxed out ram an flash with a nm-2v module for 80 bucks, paid 20 for the fxo card, got my 2900xl switch for 25 dollars, and have a 3524 poe switch coming for 50 dollars |
| 17:53.19 | bluregard | I think the polycom phones are a lot more flexible and feature rich than the ciscos |
| 17:53.34 | slav3_kitten | an most the phones i got for free in a box of cisco shit i was given |
| 17:54.18 | *** join/#asterisk odenkos (~odenkos@176.106.186.163) |
| 17:54.26 | bluregard | are they the actual cisco phones or are they the old Linksys SPA phones that are now Cisco branded? |
| 17:55.19 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 17:55.33 | slav3_kitten | they are legit cisco 7911g, 7940g, an a 7960g |
| 17:55.50 | slav3_kitten | an i'm supposedly being given 3 more 7911g's |
| 17:55.59 | bluregard | Those I haven't used so I really can't comment on them. |
| 17:56.13 | slav3_kitten | neither have lol |
| 17:57.01 | bluregard | I've been using polycoms for the better part of 10 years and can only think of a few problems I've really had with them. |
| 17:57.05 | slav3_kitten | so what kind of prices are polycom phones? |
| 17:57.55 | bluregard | I've seen IP450s for ~$150 |
| 17:58.09 | slav3_kitten | gah! *has a small heart attack* |
| 17:58.26 | bluregard | ya get what ya pay for |
| 17:59.08 | slav3_kitten | by small operation... they need a message waiting indicator light, hold button, an a few line buttons for transfering calls |
| 17:59.20 | bluregard | one of the things I really like about the polycoms that the ciscos don't have is the microbrowser. The Ciscos kind of have a "browser" but it doesn't support html. |
| 17:59.48 | slav3_kitten | yea that's like added fluff they in no way need. i'm talking like a 3 phone maximum implementation |
| 17:59.58 | slav3_kitten | with a number of extensions sending off to peoples cell phones |
| 18:04.12 | slav3_kitten | really kind of bare bones setup done cheap as possible |
| 18:05.05 | bluregard | there's always softphones |
| 18:05.24 | slav3_kitten | they want hard phones |
| 18:05.46 | bluregard | how soon are you trying to do this? |
| 18:06.36 | slav3_kitten | i'll cross that bridge when i come to it though. so for a small soho hardware wise. an older 32 bit comp with 1-2 gigs of ram and a small 50-60 gig raid decent enough for something small 20 streams max? |
| 18:06.45 | slav3_kitten | oh month or three at least |
| 18:07.54 | slav3_kitten | gotta read up, play with stuff on my end, get some confidence, show them how it works, explain it all to them, source inexpensive hardware, setup such hardware, get them a did they like an can afford. config the shit out of it |
| 18:08.43 | bluregard | all depends on the codecs that are being used. If you're not doing any transcoding that should be fine. But if you start doing a lot of conferencing with channels using all different codecs you can quickly bring a low powered machine to its knees. |
| 18:09.30 | slav3_kitten | in theory i should be able to keep everything a single codec right? |
| 18:09.44 | slav3_kitten | sip phones to pbx to sip channels? |
| 18:09.54 | bluregard | yeah |
| 18:10.10 | slav3_kitten | sorry i'm being a total noob right now |
| 18:10.37 | bluregard | its all good, I was bored anyway. |
| 18:11.24 | slav3_kitten | lol at least i'm not annoying you yet |
| 18:11.46 | slav3_kitten | i'm sure i'll have questions when i go to interface to my cisco's vic card |
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| 18:12.13 | bluregard | yeah good luck with that. I've only ever used digium cards |
| 18:12.21 | slav3_kitten | but that's a ways off. i need to get everything internal working, have someone pop under the house to run a cable to the network closet from the demarcation box outside and wire it all up |
| 18:12.34 | slav3_kitten | i've seen it can be done according to the google results |
| 18:12.50 | bluregard | I'm sure it can |
| 18:12.52 | slav3_kitten | i know i can do a skinny to pots easily within the router |
| 18:13.09 | slav3_kitten | i just need to get the book an read up on it |
| 18:13.47 | bluregard | you can download a pdf |
| 18:14.14 | slav3_kitten | gotta buy the pdf |
| 18:14.17 | bluregard | wiki.asterisk.org is also quite useful |
| 18:14.46 | slav3_kitten | my ability to have internet when i actually want to read is minimal at best. hence my sony reader |
| 18:14.47 | din3sh | bluregard: is it possible to have dynamic centralized address book with soundpoint phones? |
| 18:15.24 | bluregard | din3sh, I've seen some scripts that'll parse your voicemail.conf and convert to a polycom directory but I've never actually done it. |
| 18:15.45 | din3sh | yeah stuck with that one |
| 18:16.03 | bluregard | din3sh, its been in the back of my mind though. I might tackle it after this parking thing is finished. |
| 18:16.04 | din3sh | the directory is local rather than on server side |
| 18:16.22 | bluregard | you can put a directory on your provisioning server |
| 18:16.31 | din3sh | phone downloads a static xml file |
| 18:16.38 | bluregard | right |
| 18:16.47 | din3sh | i asked polycom about it |
| 18:17.00 | din3sh | they say it can work with an ldap server |
| 18:17.16 | din3sh | but that requires you to purchase some kind of licence to get it working |
| 18:17.39 | din3sh | i think that might be the MAC-licence.cfg file the phone requests |
| 18:18.04 | bluregard | one thing you could do is write a cgi or php script or whatever that you access from the polycom microbrowser that either parses voicemail.conf and creates tel:// urls out of all of the entries or maybe use AMI to pull down extensions. |
| 18:18.54 | bluregard | then re-map your directory button to connect directly to that cgi/php script. |
| 18:19.26 | din3sh | so far i can make a php file write in the xml file, but the phone has to be rebooted to download the updated xml directory |
| 18:19.50 | bluregard | right, but using the microbrowser you can do it in realtime |
| 18:19.58 | din3sh | hmm |
| 18:20.07 | din3sh | will give that a try |
| 18:20.35 | bluregard | the polycom can dial tel://<num> links by selecting them and pressing the ok button. |
| 18:20.55 | din3sh | grandstream has same kind of static xml file, except with grandstream you can setup the phone to periodically download the xml file |
| 18:21.25 | bluregard | I think I've heard grandstream phones described as the yugo of voip |
| 18:22.14 | din3sh | are they that bad? |
| 18:22.35 | bluregard | I have to say, I've never heard a good thing said about them, but to be fair I've never used one personally. |
| 18:23.28 | bluregard | I want to say that the polycom can periodically download your directory too but I'm not 100% on that one. |
| 18:24.21 | bluregard | how often is your directory going to change though |
| 18:25.02 | din3sh | anytime a user makes a change in sip realtime table |
| 18:25.06 | din3sh | ;) |
| 18:25.41 | din3sh | i am testing granstream phone actually |
| 18:25.54 | din3sh | i have seen worst |
| 18:26.06 | din3sh | atcom phones |
| 18:26.57 | bluregard | well if you're using realtime then I would definitely use a cgi/php script from the microbrowser. Pull down all sip entries from your realtime table and build tel:// URIs out of them. Problem solved. |
| 18:28.07 | din3sh | can polycom's mircobrowser do that? |
| 18:29.01 | bluregard | well the microbrowser would just be requesting the cgi/php script that does the actual work then just displays the results to the microbrowser. |
| 18:29.33 | din3sh | ok will look in that direction |
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| 18:33.14 | bluregard | I have a php script that gets all parked calls via AMI and builds tel:// urls out of them so users can just scroll down the list and click on the one they want to dial the parked ext and pick up the call. |
| 18:40.30 | slav3_kitten | oh bluregard one dumb question before i take off for the day. i can have multiple dids and setup a dial plan to place the right dial sequence to the right place right? like a us did, canadian did, united kingdom did. |
| 18:40.46 | din3sh | stupid question, how do we make the phone run the php script? |
| 18:41.46 | [TK]D-Fender | slav3_kitten, Sure |
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| 18:42.34 | din3sh | bluregard:how do we make the phone run the php script? |
| 18:48.21 | bluregard | slav3_kitten, yes |
| 18:49.22 | bluregard | din3sh, you either need to set your home page for the microbrowser to the script or to an html page with a link to the script or you should be able to remap your directory button to go directly to the script probably with EFK. |
| 18:51.18 | bluregard | din3sh, you could also set up a softkey to do it but that would be kinda redundant |
| 18:53.50 | din3sh | thanks bluregard, really helpfull |
| 18:54.04 | bluregard | din3sh, no problem |
| 18:54.10 | din3sh | was about to buy the licence for the ldap |
| 18:54.12 | din3sh | lol |
| 18:56.24 | bluregard | yeah nuts to that |
| 18:56.42 | bluregard | how much is the ldap license anyway |
| 18:56.55 | din3sh | polycom havent replied yet |
| 18:57.04 | bluregard | shocker |
| 18:58.04 | din3sh | yeah, in 2012 we have a phone that cannot have a centralized directory the simple way |
| 18:58.14 | din3sh | and they dont come that cheap either |
| 18:58.53 | bluregard | nothing ever does |
| 18:59.00 | din3sh | lol |
| 18:59.55 | din3sh | have you had experience with polycom hdx video conferencing? |
| 18:59.58 | [TK]D-Fender | LDAP phonebook isn't licensed on the Polys any more |
| 19:00.16 | bluregard | I have not |
| 19:00.32 | bluregard | only video I've done through asterisk has been with softphones |
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| 19:01.41 | din3sh | i came accross a company reselling the system @ $20k |
| 19:02.05 | mlerley | hi folks |
| 19:04.56 | din3sh | so the microbrowser is configured at [microbrowser mb.proxy= ""] in the config file |
| 19:05.09 | mlerley | I'm trying a new install of AsteriskNOW with a digium AEX800 card with 6 FXO ports. I can't get incoming or outgoing calls; I don't even see the incoming call data in the CLI. I'm thinking it must be something simple re: channels, etc. Thoughts? |
| 19:10.30 | bluregard | din3sh, no, hang on |
| 19:11.03 | bluregard | mb.main.home= |
| 19:11.56 | bluregard | you can also define a softkey to pull down something to the MB by doing softkey.n.action="http.domain.com" |
| 19:13.09 | din3sh | in which file is the softkey... =? |
| 19:13.36 | bluregard | applications or features I think |
| 19:14.05 | bluregard | softkeys are in features |
| 19:14.05 | wonko | hmm, calls go out to google but they don't come back in. interesting. |
| 19:15.35 | din3sh | ok |
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| 19:17.56 | csd-199 | Hi. Is it possible to install asterisk 1.8 on centos 6 using yum? |
| 19:19.27 | wonko | csd-199: you'd have to find a repo, it's not in any of the default repos |
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| 19:23.52 | mlerley | ah! got it. missing the include of dahdi-channels.conf in chan_dahdi.conf. no idea why that wasn't there after installing asterisknow |
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| 19:26.22 | saliak | I'd like to enable fax (incoming and outgoing) support on my "Asterisk 1.6.2.5-0ubuntu1.3" install. I installed this out of apt a while back. anyone know of a good guide to walk me through how to do this? most i've found reference installing spandsp prior to compiling and installing asterisk from source |
| 19:26.54 | malcolmd | mlerley: please contact our support department - +1 256 428 6000 or http://www.digium.com/en/users/support-create-a-case/ they'll be happy to have a look |
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| 19:32.31 | mlerley | malcolmd: thanks. i still have another issue i may contact you about. |
| 19:33.07 | malcolmd | generally, i'll defer to the Support department. any time a digium product for which you paid money is involved, our Support department should be your destination. :D |
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| 19:49.02 | bluregard | din3sh, did you get the MB working? |
| 19:53.23 | saliak | OK, looks like ReceiveFax is included so all set. Now i'm having issues actually receiving a fax (fails on comm errors). Is there a way to slow down the connection or something to make this more robust? |
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| 20:13.18 | fubada | is there an issue specific to sip providers and 1866 numbers, im having trouble dialing my 1-866 did from within my own pbx |
| 20:13.24 | fubada | which uses that provider for general sip outgoing |
| 20:14.15 | talntid | no, there is not |
| 20:14.18 | talntid | what trouble are you having? |
| 20:15.01 | fubada | a call to 1-866 number that is owned by my provider from within my pbx doesnt complete |
| 20:15.08 | fubada | a call to a random 1-866 number works |
| 20:15.19 | fubada | a call from my 1-866 number from my cellphone also works |
| 20:15.26 | talntid | if you call that 166 # from a cell phone or something, does it complete? |
| 20:15.28 | talntid | oh, ok |
| 20:15.31 | fubada | yes it does |
| 20:15.48 | talntid | is your outgoing callerid your same 1-866-# ? |
| 20:15.54 | talntid | from within your pbx |
| 20:16.00 | fubada | no |
| 20:16.34 | talntid | when you say it doesn't complete... you see * push the call outgoing, but then you just don't see the incoming... or you see the incoming improperly, or what? |
| 20:17.02 | fubada | theres more to this, the 1-866 number is hosted by a different asterisk box (1.6), on a different IP. Im trying to call this number from my new asterisk box (10.4.2) which is on another IP |
| 20:17.07 | fubada | through the same provider |
| 20:17.14 | fubada | and i think it doesnt work for 1-866 calls |
| 20:17.33 | fubada | i see in * logs it establishes a session with my provider, then silence, then nothing |
| 20:18.04 | fubada | heh nvermind this is too confusing to explain |
| 20:18.21 | fubada | thanks |
| 20:18.23 | talntid | but when, from your cell phone, you call the 1-866-#, it shows up on the 10.4.2 box? |
| 20:18.49 | fubada | no, it shows up on the 1.6 box, and this is correct |
| 20:18.51 | fubada | as it should |
| 20:18.55 | talntid | gotcha |
| 20:19.34 | fubada | i think its something specific to 1-866 number routing on my sip provider side |
| 20:19.40 | fubada | im calling them to check on this |
| 20:19.49 | talntid | yeah, I'd call ITSP too |
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| 20:19.54 | fubada | this isnt really * related, thanks for the help though ;) |
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| 20:44.25 | mlerley | ah... fixed it! format of dahdi-channels.conf was wrong. had extra callerid/group/context lines for each channel with empty entries that overrode the previously specified correct ones |
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| 21:14.13 | bluregard | is anyone having problems with inbound google voice? |
| 21:14.34 | bluregard | it was working the other day but doesn't appear to be working at the moment |
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| 23:00.19 | adeel | stupid switchvox.... |
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| 23:47.07 | pzn | I have one asterisk server with extensions from 700 to 799, and other asterisk server with extensions 800 to 899. how can I connect both asterisks servers? any howto? |
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| 23:49.46 | pzn | what is the name of this feature that I need, so I can search google? |