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| 00:56.31 | WIMPy | Is that a thing of its own or an add-on package? |
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| 02:14.50 | jeffspeff | where can i find current information on using extensions in a db? all of the info i can find showing db tables, etc. is all on voip-info.org which (as i understand) is completely antiquated. |
| 02:21.38 | WIMPy | Not always, but often. |
| 02:22.09 | WIMPy | But then I don't see what should have changed there. |
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| 02:27.28 | jeffspeff | WIMPy, i'm not sure what would have changed either, but i start seeing references to asterisk 1.6 and i'm using 11.2, i get kinda cautious |
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| 02:40.50 | [TK]D-Fender | extensions.conf hasn't really changed |
| 02:40.55 | [TK]D-Fender | So that spec needn't |
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| 03:36.26 | jeffspeff | [TK]D-Fender, can you still use dialplan variables when using extension db switches? |
| 03:41.47 | [TK]D-Fender | ? |
| 03:44.05 | jeffspeff | nvm, i found it |
| 03:44.32 | jeffspeff | i was looking to see if you could still use things like ${EXTEN} when using realtime extensions. |
| 03:44.43 | jeffspeff | i didn't know how it would be interpreted. |
| 03:47.51 | fling | How to call tox users? |
| 03:48.11 | fling | tried to google but got this in the results: |
| 03:48.13 | fling | remove asterisk with yum [Archive] - FedoraForum.org |
| 03:48.32 | fling | [TK]D-Fender: Hello. :> |
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| 09:22.11 | linocisco | we have the office with Panasonic PABX KX-TES824 with 2 CO lines, and anolog phones configured. We want to give voice mail and IVR and other advanced options using asterisk. Panasonic has only USB port like printer to connect to Computer and RS-232C male port |
| 09:23.12 | linocisco | and want to use IP phone as well. the thing is I do not want to buy FXO asterisk cards, I would like to use only FXO interface (CO line ports) on Panasonic |
| 09:24.07 | Chainsaw | linocisco: To do that the Panasonic would have to speak SIP and contain an ethernet port. Is that the case? |
| 09:25.40 | linocisco | Chainsaw, need to interface asterisk and Panasonic, using USB port or RS-232C |
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| 09:25.52 | linocisco | Chainsaw, is there anyway? |
| 09:25.59 | Chainsaw | linocisco: Doubtful. |
| 09:26.23 | Chainsaw | linocisco: If the USB port only implements line printer protocol, that's a dead end. Shame, as it would have had the bandwidth to carry the calls. |
| 09:26.29 | linocisco | somebody has done same setup without buying FXO asterisk cards? |
| 09:26.43 | Chainsaw | linocisco: If they have, they're keeping awfully quiet... |
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| 09:46.51 | WIMPy | Maybe you can buy a VOIP card for the Panasonic. But that might be even more expensive. |
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| 12:15.14 | jzu_ | does this error message cripple SIP or does resolving hostname succeed if I dont see it anymore after first error? |
| 12:15.17 | jzu_ | [2015-05-05 10:27:32] NOTICE[1795] chan_sip.c: Warning: Re-lookup of 'A4C3146668A0292DD24B94059433A3744.asuscomm.com' failed! |
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| 12:46.14 | Penguin | jzu_: That's not an error. |
| 12:47.13 | Penguin | I think, for the most part, warnings are not fatal. |
| 12:48.32 | jzu_ | Yeah, could be so :) |
| 12:48.55 | jzu_ | if chan_sip still has the IP cached then it's just informational |
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| 13:37.55 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 13:38.59 | EmleyMoor | When Asterisk is processing a regular expression, does it use "capturing groups" and would it be more efficient to make them "non-capturing"? |
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| 13:42.53 | wdoekes | EmleyMoor: where? |
| 13:43.29 | EmleyMoor | wdoekes: Using a =~ or : operator in the dialplan |
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| 13:45.06 | mr_hindi | hey |
| 13:45.14 | wdoekes | ./main/ast_expr2.c says regcomp gets no REG_NOSUB |
| 13:45.26 | wdoekes | but I doubt that you'd notice any difference if you added that |
| 13:45.36 | wdoekes | unless you run =~ in a very very tight loop |
| 13:45.53 | EmleyMoor | wdoekes: So it doesn't actually use capture groups, or it doesL |
| 13:45.54 | EmleyMoor | ? |
| 13:48.19 | wdoekes | EmleyMoor: actually, it does use the outermost parentheses |
| 13:48.22 | wdoekes | http://wjd.nu/notes/2011#asterisk-dialplan-peculiarities-regex |
| 13:48.51 | mr_hindi | hey. when i do an outbound call my asterisk 1.8 shows "requested media update control 26" many times and after that ,the voice quality is very bad... |
| 13:49.05 | mr_hindi | any ideas? |
| 13:49.24 | mr_hindi | If I hangup and place the call a second time everything is fine |
| 13:49.58 | mr_hindi | I can't reproduce that.. sometimes its happens sometimes not... |
| 13:50.35 | EmleyMoor | Is using the : operator deprecated? |
| 13:50.43 | wdoekes | AST_CONTROL_SRCCHANGE = 26,/*!< Media source has changed and requires a new RTP SSRC */ |
| 13:51.08 | wdoekes | are you using g729 when it sounds shitty? |
| 13:51.36 | mr_hindi | nope, only g711a and u are allowed |
| 13:51.45 | wdoekes | EmleyMoor: not officially, afaik |
| 13:52.03 | wdoekes | but I believe it anchors on one side, which you can do manually with =~ if you want |
| 13:53.19 | mr_hindi | wdoekes, i've checked the siptrace the sdp shows alaw as expected... |
| 13:54.02 | mr_hindi | I spent about 3 days on this but I dont get it... |
| 13:55.26 | wdoekes | mr_hindi: sorry, I don't have a quick rtp debug tool for you. my first suggestion would be to open up the entire audio stream in wireshark and see if you can see anything out of the ordinary there |
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| 13:57.47 | wdoekes | mr_hindi: or it would be that someone else is unintentionally breaking into your stream |
| 13:57.57 | wdoekes | do you have strictrtp enabled (rtp.conf)? |
| 13:59.22 | mr_hindi | wdoekes, strictrtp isn't enabled |
| 14:00.10 | mr_hindi | I've spent hours in rtp stream analysis... |
| 14:00.19 | wdoekes | enable strictrtp if you can |
| 14:00.21 | mr_hindi | but can't find any errors.. |
| 14:00.41 | mr_hindi | wdoekes, ok I will try |
| 14:01.21 | wdoekes | mr_hindi: if you did rtp stream analysis you would know whether: (a) the ssrc keeps changing back and forth, or only forth |
| 14:01.36 | wdoekes | (b) there is a third IP address in the mix |
| 14:02.13 | mr_hindi | can't find a another ip adress in the stream |
| 14:02.14 | wdoekes | (c) there is double the traffic (a second stream), or the same amount |
| 14:02.47 | mr_hindi | I've started tcpdump.. now I have to wait for it.... |
| 14:03.25 | mr_hindi | Will check your suggestions |
| 14:03.29 | mr_hindi | thanks |
| 14:04.28 | EmleyMoor | When I construct a regexp, a "non-capture group" has ?: at the beginning. Does Asterisk take any notice either way of those? |
| 14:04.34 | mr_hindi | maybe I've missed something |
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| 14:06.41 | wdoekes | EmleyMoor: ?: is not EXTENDED_REGEX, so regcomp should return REG_BADRPT |
| 14:07.52 | wdoekes | but why are you so focused on tuning the =~ operator? |
| 14:09.54 | EmleyMoor | wdoekes: I'm not - but I've got two rather fiendish-looking regexps that I'm looking to make as compact as possible. |
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| 14:31.00 | EmleyMoor | A bizarre 1 character difference between two seemingly identical regexps... |
| 14:35.47 | EmleyMoor | Is there a list anywhere of what's deprecated in Asterisk 11 (and for that matter 13)? |
| 14:37.18 | [TK]D-Fender | CHANGES <- should list them |
| 14:37.57 | EmleyMoor | Also, how do I see which modules I am using? |
| 14:39.44 | [TK]D-Fender | "modules show" |
| 14:40.05 | [TK]D-Fender | think that should be singular... |
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| 14:41.06 | wdoekes | that lists what you've got loaded, not necessarily what you're using. there is no command for that, alas |
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| 14:44.47 | EmleyMoor | wdoekes: Well, it answered my question anyway - using xmpp rathder than the deprecated jabber module |
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| 14:48.47 | overyander | can i have more columns in an realtime extension table than specified in http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ? |
| 14:50.13 | [TK]D-Fender | Pretty sure that extras won't matter.... * simply shouldn't use them for anything |
| 14:51.42 | overyander | that's what i was thinking too, just wanted to double check... thanks |
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| 14:59.02 | overyander | can you have extens in a context flat file and then switch to the db at the end of the extens? |
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| 15:02.43 | vomit | Hey. I have a Cisco 7940g, I reconfigured it's tftp config file to use different extension and rebooted the phone, the phone picked up new config and when I telnet and show config - I see it is updated correctly, however it registers now using the old extension it had and new extension despite the fact old extension is not in the config. Any tips? |
| 15:05.30 | [TK]D-Fender | overyander: IIRC no. You'd make a containing context, then include one with your fixed, then another with the switch |
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| 15:09.14 | EmleyMoor | I would set up Google Talk capability on my system, but want to know how it might interact with the gateway I'm using to bring the messaging to my own XMPP server |
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| 17:44.30 | qakhan | i am tryig to change Caller id on incoming call, it is not change but new caller id attach with actuall caller id |
| 17:45.50 | [TK]D-Fender | Now try rephrasing that into a clearer sentence |
| 17:46.20 | [TK]D-Fender | And SHOW us what you're doing and what is actually happening |
| 17:46.28 | robmal | I think he wants xxx to become xxxyyy |
| 17:48.18 | qakhan | here is pastbin http://pastebin.com/GnAGFCpF |
| 17:50.10 | qakhan | when user receive incoming call they should not see actuall Caller id. so i am changing caller id on incoming call. |
| 17:50.19 | [TK]D-Fender | I don't see SIP DEBUG included there to prove what came in VS what went out |
| 17:50.54 | qakhan | i can see changed caller id but actuall caller id comes with it. |
| 17:51.01 | qakhan | ok i send you sip debug |
| 17:54.25 | qakhan | here is sip debug http://pastebin.com/0qP19BAu |
| 17:55.46 | [TK]D-Fender | Cam in as : From: "+12404835218" <sip:2404835218@66.241.99.83>;tag=as1f3024d6 |
| 17:55.56 | [TK]D-Fender | went out as : From: "+12404835218" <sip:12345678@192.168.10.26>;tag=as2cb078ed |
| 17:56.16 | [TK]D-Fender | So it did exactly what you told it to. |
| 17:56.31 | [TK]D-Fender | -- Executing [s@regencyoutgoing:1] Set("SIP/vitel-inbound-000000da", "CALLERID(num)=12345678") in new stack |
| 17:57.11 | qakhan | that is fine, but i dont want to show +12404835218 to user |
| 17:57.19 | [TK]D-Fender | sO GO CHANGE IT |
| 17:57.30 | qakhan | how? |
| 17:57.35 | [TK]D-Fender | ... |
| 17:57.40 | [TK]D-Fender | "core show function CALLERID" <- |
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| 18:01.01 | RoyK | hm... -- Error initializing Bluetooth device RoysSGS5. |
| 18:01.11 | RoyK | any idea what might be the problem? |
| 18:02.51 | qakhan | which item in CALLERID will change it> |
| 18:03.41 | *** join/#asterisk ghostlines (~ghostline@i191002.upc-i.chello.nl) |
| 18:19.10 | [TK]D-Fender | qakhan: .... THE NAME |
| 18:19.47 | RoyK | anyone here using chan_mobile? it doesn't seem to want to talk to my phone... |
| 18:22.50 | qakhan | [TK]D-Fender thanks, i forgot CALLERID(name) |
| 18:23.12 | qakhan | it is working now |
| 18:24.22 | *** join/#asterisk y_ateya (~y_ateya@41.45.42.254) |
| 18:25.24 | y_ateya | Hi, how to enable jitter buffer for PJSIP? in normal sip.conf, I would say jbenable=yes. No similar configuration for pjsip.con |
| 18:27.02 | RoyK | seems chan_mobile is trying to connect - repeatedly - repeatedly - http://paste.debian.net/171606/ |
| 18:27.16 | RoyK | any idea how to find out what goes wrong? |
| 18:28.25 | RoyK | nothing in the logs... |
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| 18:35.29 | rmudgett | y_ateya: Read up on func JITTERBUFFER |
| 18:36.01 | y_ateya | so I have to enable it using JITTERBUFFER function in the dialplan, not through configuration file like sip? |
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| 18:38.50 | RoyK | generic jitterbuffer? nice |
| 18:38.56 | dtrainor | Good afternoon. When using GoSub, is it safe to call the same context from which you executed GoSub, or should I always call a different context? |
| 18:39.08 | RoyK | remembers posting the original jitterbuffer code to asterisk some eight years back |
| 18:41.23 | file | y_ateya, there is a set_var option on endpoints which allows the setting of dialplan variables or dialplan functions on a channel |
| 18:43.36 | RoyK | file: long time no see :P |
| 18:43.43 | file | hola |
| 18:45.05 | RoyK | is trying and so far failing to make chan_mobile work :( |
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| 18:49.31 | y_ateya | BTW, how bad is to two multiple jitter buffers in the same path? will it just insert a delay OR it will confuse the 2nd jitter buffer? |
| 18:49.51 | RoyK | y_ateya: probably quite bad |
| 18:50.37 | RoyK | y_ateya: that is - where in the path are the two? a jitterbuffer shouldn't merely insert a delay - it should fix jitter |
| 18:53.52 | y_ateya | Many times I run in these scenarios: SIP-caller -> SIP-server-1 -> SIP-server-2 -> SIP-callee, SIP-caller -> SIP-server-1 -> SIP-callee, SIP-caller -> SIP-server-2 -> SIP-callee, and jitter buffer on SIP-callee is really bad. So I turn it on SIP-server-1 and SIP-serve-2. |
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| 18:56.27 | RoyK | y_ateya: normally the jitterbuffer should only be used at endpoints |
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| 19:06.52 | cleb | Seeing a weird issue, Asterisk 1.8.19.0. I am using Digium Cloud Service as my SIP provider, with Cisco 7905's with a SIP load. When I make calls from the 7905 out over the DCS trunk, I won't have any recieve audio on the 7905 until I talk into the phone or make a noise, then suddently the RX audio is live? Any ideas? |
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| 19:13.03 | y_ateya | cleb: Do you have silence suppression activated in any part of this chain? |
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| 19:19.10 | y_ateya | cleb: I read that Cisco 7905 has silence suppression enabled by default (http://www.voip-info.org/wiki/view/Cisco+7905/7912+IP+Phones). Normally this is the source of no-reception-until-first-real-voice. |
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| 19:22.33 | cleb | y_ateya: I will look into that, thank you! |
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