| 00:06.55 | *** join/#asterisk Qwell (north@pdpc/sponsor/digium/Qwell) |
| 00:06.56 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:11.28 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
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| 00:58.49 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 01:24.26 | raspberrypifan | does asterisk work as a softswitch? |
| 01:25.20 | [TK]D-Fender | no. * is a ... |
| 01:25.23 | [TK]D-Fender | ~b2bua |
| 01:25.23 | infobot | well, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 01:25.50 | raspberrypifan | hmm is there an opensource softswitch? |
| 01:27.41 | [TK]D-Fender | Perhaps you should be very sure of your use of the terms as well as your actual needs.... |
| 01:29.24 | raspberrypifan | well this is for my uncle he apparently is trying to use something called IpSwitch which seems to be to very limited and expensive |
| 01:31.39 | raspberrypifan | i know very little of course but im here to learn |
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| 01:35.11 | raspberrypifan | i hope i didnt miss out on anything |
| 01:35.23 | [TK]D-Fender | nope |
| 01:36.04 | raspberrypifan | were u provinding me with an answer |
| 01:36.36 | [TK]D-Fender | Not aware of anything that counts as a solid class4/5 SS |
| 01:36.42 | [TK]D-Fender | that's OSS |
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| 01:37.23 | [TK]D-Fender | Then again.... who says that's the actual kind of product required? |
| 01:37.32 | [TK]D-Fender | unclear need = loaded question |
| 01:38.52 | raspberrypifan | well he seems to be doing something with call termination and buying and selling minutes |
| 01:38.56 | raspberrypifan | to TATA commnications |
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| 01:44.40 | raspberrypifan | im trying to understand his business, because he seems to want to bring me on board. He's got a CCNP but i know more |
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| 03:20.31 | volga629 | Hello Everyone, I am looking information about this error ERROR[4658]: tcptls.c:96 ssl_close: SSL_shutdown() failed: 5 |
| 03:26.23 | *** join/#asterisk songoku1610 (~root@123.29.67.13) |
| 03:26.49 | songoku1610 | Hi |
| 03:27.04 | songoku1610 | I used voiceglue with asterisk |
| 03:27.23 | songoku1610 | but I face problem when use transfer tag |
| 03:27.31 | songoku1610 | <transfer name="MyCall" dest="tel:+11112223333" bridge="true" connecttimeout="20s"> |
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| 03:28.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:28.15 | songoku1610 | when called party don't answer, or hungup, but callee not hungup |
| 03:28.32 | songoku1610 | are there anyone know how to resolve it? |
| 03:28.40 | songoku1610 | please help me, thank you so much |
| 03:31.10 | raspberrypifan | whats a voice glue |
| 03:31.59 | *** join/#asterisk Dovid (~Dovid@ool-2f113d03.dyn.optonline.net) |
| 03:33.44 | songoku1610 | voice glue is a opensource, that is a agi server ,it build vxml to dialplan |
| 03:36.27 | [TK]D-Fender | That is not something I've ever heard mentioned in here, nor is it something that we support. |
| 03:58.49 | volga629 | what possible cause for this error [2014-05-30 05:57:07] ERROR[7766]: tcptls.c:96 ssl_close: SSL_shutdown() failed: 5 |
| 03:58.49 | volga629 | SSL certificate ok |
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| 05:01.07 | snadge | i have had a few customers try registering to us on ports other than 5060 .. but it still shows as 5060 in the via header |
| 05:01.37 | snadge | i dont know if thats potentially a known bug, or if anyone knows any caveats with using a udp port other than 5060 |
| 05:02.04 | [TK]D-Fender | * will send to where it's told to send to. |
| 05:05.46 | snadge | the register string correctly shows the 5065 or whatever... but what does this mean "Defaddr->IP : 0.0.0.0 Port 5060" |
| 05:07.41 | [TK]D-Fender | Means it hasn't registered |
| 05:08.21 | [TK]D-Fender | Sorry, wrong field |
| 05:08.37 | [TK]D-Fender | that's a default to tr in the case it HASN'T registered |
| 05:09.00 | [TK]D-Fender | Typically you should see it in Addre->IP |
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| 07:25.50 | hellinterim | Hi guys. I'm looking to buy a couple SNOM phones. Maybe only 5. Is there a good place to buy these or should I just keep my eye out on ebay and similar sites? |
| 07:26.26 | hellinterim | I normally only ask about asterisk related topics. Feel free to tell me off if this is really offtopic. |
| 07:29.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:33.18 | *** join/#asterisk KOPRajs (52638852@gateway/web/freenode/ip.82.99.136.82) |
| 07:34.10 | KOPRajs | hi, is it possible to set something like alias to SIP peer in Asterisk? |
| 07:35.18 | KOPRajs | I've got a client with configuration I can't change which registers with name and username 1234 |
| 07:36.06 | KOPRajs | so I have to create SIP account with that parameters |
| 07:36.35 | KOPRajs | and it can be then called like SIP\1234 from my dialplan |
| 07:37.04 | KOPRajs | but I need it to be callable by SIP\xyz1234 |
| 07:54.06 | bulkorok | KOPRajs: exten => 123,1,Dial(SIP/xyz1234) dials xyz1234 when 1234 is called |
| 07:55.52 | snmp | 1234 |
| 07:56.16 | snmp | exten => 1234,1.. |
| 07:56.40 | bulkorok | sureā¦ typo |
| 07:57.29 | KOPRajs | I know this, but I've got some more complex dialplan which expects SIP\xyz1234 for many other clients |
| 07:58.34 | *** join/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 07:58.40 | Zogot | morning |
| 07:59.11 | ChannelZ | What you don't own a text editor with search and replace? |
| 07:59.22 | *** join/#asterisk timahvo1 (~rogue@41.215.138.203) |
| 07:59.26 | KOPRajs | and now I'm going to get few clients with different configuration (without xyz) and all these will come through the same SIP trunk as those with xyz |
| 08:00.37 | KOPRajs | I'm able to distinguish between them on incoming call |
| 08:00.43 | KOPRajs | *I'm not abl |
| 08:01.12 | KOPRajs | so I would like to set an alias with xyz for those without it |
| 08:04.40 | ChannelZ | The easiest thing to do probably is to make a macro or something which figures out what the hell it's supposed to do based on whatever odd logic you have going |
| 08:05.06 | KOPRajs | the problem is that there is no logic |
| 08:05.13 | ChannelZ | clearly |
| 08:06.08 | KOPRajs | it will be simply many random numbers coming through one SIP trunk from provider (so one incoming context) |
| 08:06.48 | KOPRajs | some of them callable as SIP\xyz1234 and others as SIP\1234 where 1234 is the incoming number |
| 08:08.20 | KOPRajs | what is worse, the numbers are really random as there are numbers transferred from PTSN so I can't even distinguish between them based on the number |
| 08:09.20 | *** part/#asterisk LiuYan (~LiuYan@58.251.146.232) |
| 08:10.03 | ChannelZ | So how the hell is it going to know what peer to dial in the first place? I don't know how this "alias" helps you. You're basically wanting to be lazy and Dial(SIP/${EXTEN}) with no idea what ${EXTEN} even is. |
| 08:10.41 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 08:11.41 | KOPRajs | I would like to set something like alias in the sip section for those new numbers... so they are accessible by both SIP\1234 and SIP\zxy1234 |
| 08:13.20 | *** join/#asterisk Draecos (~Draecos@106-69-174-3.dyn.iinet.net.au) |
| 08:13.42 | KOPRajs | the simple solution would be to create the new accounts the same way as the existing ones (with the xyz before number) but that would envolve physically change configuration on many clients which is impossible |
| 08:14.19 | ChannelZ | what is "xyz" derived from? |
| 08:14.31 | KOPRajs | basically they are configured for different provider right now and they are going to be transferred by simply changing the DNS to my server |
| 08:14.59 | KOPRajs | xyz is static string |
| 08:15.24 | KOPRajs | just badly choosen naming pattern in history |
| 08:15.50 | ChannelZ | and why is it you can't just make your dialplan dial the peer? |
| 08:16.11 | ChannelZ | dial the right peer that is |
| 08:17.05 | KOPRajs | there are existing peers with the xyz and they are going to be these new without xyz all incoming through the same incoming connection (context) |
| 08:17.25 | ChannelZ | so? |
| 08:17.27 | KOPRajs | all random numbers |
| 08:18.20 | KOPRajs | I want to avoid making a rule for every number |
| 08:19.03 | KOPRajs | I'll do that if there is no other way of course |
| 08:20.01 | ChannelZ | What would the "rule" be? You said earlier there is no logic that can be derived. I feel like maybe you're confused about extensions vs device names. |
| 08:20.43 | ChannelZ | I don't understand how you can simultaneously say that calls will come in to random extensions, but somehow you dial an appropriate peer depending on it. |
| 08:21.05 | KOPRajs | exten => _[0-9].,1,Dial(SIP/xyz${EXTEN:0},60,Tt) |
| 08:21.16 | KOPRajs | there is something like this right now |
| 08:21.55 | KOPRajs | so there is no need for changing the dialplan if new peer is added to sip.conf |
| 08:24.38 | KOPRajs | something similar is on outgoing call but I can put those new numbers in different context and copy the dialplan with xyz removed |
| 08:25.08 | ChannelZ | You could use something like ChanIsAvail to see if xyz1234 exists, and if not dial without it. |
| 08:25.34 | KOPRajs | that sounds like it could help |
| 08:25.48 | KOPRajs | how do I check that? |
| 08:26.18 | ChannelZ | or the SIPPEER function |
| 08:27.12 | ChannelZ | Well use GotoIf or something to look at the result of the test and respond accordingly.. |
| 08:28.00 | KOPRajs | will look at it |
| 08:28.03 | KOPRajs | thank you |
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| 08:30.57 | KOPRajs | what would be the safe way of testing wherher the peer exists? |
| 08:31.13 | KOPRajs | exten => s,1,Set(foo=${SIPPEER(thatguy,context)}) |
| 08:31.15 | KOPRajs | ? |
| 08:31.45 | KOPRajs | sorry, I meant just ${SIPPEER(thatguy,context)} |
| 08:32.17 | KOPRajs | if this is empty? |
| 08:37.02 | ChannelZ | Well let me think here, the easiest way given your setup.. |
| 08:39.28 | KOPRajs | I'm thinking something like GotoIf(${SIPPEER(xyz${EXTEN:0},context)}<>NULL,xyz) not sure about the syntax but that's not problem |
| 08:41.01 | KOPRajs | exten => _[0-9].,n,Dial(SIP/${EXTEN:0},60,Tt) |
| 08:41.08 | KOPRajs | exten => _[0-9].,xyz,Dial(SIP/xyz${EXTEN:0},60,Tt) |
| 08:42.03 | KOPRajs | just not sure if the test is safe |
| 08:44.44 | *** join/#asterisk danjenkins (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 08:49.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 08:49.13 | ChannelZ | sorry sidetracked on something else |
| 08:50.14 | ChannelZ | The cheesiest way to do it is to say Set(MyPrefix=${IF(${ISNULL(${SIPPEER(${EXTEN})})}=1?xyz)}) and then Dial(SIP/${MyPrefix}${EXTEN}) |
| 08:50.34 | ChannelZ | Then you don't have to bother with gotos and labels |
| 08:51.35 | ChannelZ | Basically it says If SIP/${EXTEN} doesn't exist (IE without the xyz), set ${MyPrefix} to xyz |
| 09:02.49 | Zogot | still having the issue where the registered contact and the def. username dont match |
| 09:03.03 | Zogot | ive seen the register packet and it is valid |
| 09:03.16 | Zogot | except perhaps the uuid? not sure how that plays into things |
| 09:03.35 | KOPRajs | thanks, that seems nicer than goto :-) |
| 09:06.23 | *** join/#asterisk snmp (~snmp@81.95.129.133) |
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| 09:16.25 | Zogot | what is urn:uuid:00000000-0000-1000-8000-00085D10183A in REGISTERs? |
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| 10:31.37 | Zogot | im really lost on this wrong defined username peer issue |
| 10:32.13 | Zogot | the packet doesn't show anything odd, the only thing I can think of was the uuid reference, wether that could cause these complications |
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| 10:54.46 | Geek-Linux | Hi. All i am trying video call on asterisk and gets this message "chan_sip.c:10215 process_sdp: Ignoring video stream offer because port number is zero" |
| 10:55.16 | Geek-Linux | can any one of you help me in this regards. |
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| 11:00.01 | Geek-Linux | Hi. All i am trying video call on asterisk and gets this message "chan_sip.c:10215 process_sdp: Ignoring video stream offer because port number is zero" can any one of you help me in this regards. |
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| 11:09.42 | Geek-Linux | <PROTECTED> |
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| 11:37.51 | Zogot | can anyone explain this: https://gist.github.com/zogot/19d5c9863974607383ee |
| 11:38.03 | Zogot | line 51 is the IP of the intersport extension |
| 11:40.42 | r00f | not sure that i understand the question |
| 11:41.25 | r00f | from the looks of it, your phone is behind nat, and line 51 is its public ip. or what? |
| 11:46.54 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
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| 11:50.36 | bulkorok | what does EXTEN:-1 exactly does?! |
| 11:50.52 | bulkorok | only the firts "right" digit? |
| 11:51.08 | bulkorok | so if exten is 1234 exten:-1 is just 4 ?! |
| 11:52.50 | *** join/#asterisk rhineheart_m (~chatzilla@unaffiliated/rhineheartm/x-283746) |
| 11:53.19 | Zogot | r00f: the issue is the registered information for the phone doesnt match what the user has specified |
| 11:53.27 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
| 11:53.35 | bulkorok | yeah it is.,.. |
| 11:53.40 | Zogot | so a call from chasex2d-202 to chasex2d-203 goes to intersportveldhuis-212 |
| 11:53.58 | Zogot | https://gist.github.com/zogot/19d5c9863974607383ee |
| 11:54.18 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 11:56.08 | Zogot | it seems to cycle between 3/4 of wrong registered info |
| 11:56.15 | Zogot | yesterday it was going to a different client |
| 11:57.12 | Zogot | https://gist.github.com/zogot/cf327bcdb2121d6ccfd0 was an example yesterday, today its now https://gist.github.com/zogot/f1b1d994aff9ae289395 |
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| 12:00.40 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 12:01.34 | dan_j | Hi. leifmadsen mentioned that asterisk 12 handle recordings and transfers a little better than 11. How suitable is 12 for a production environment? |
| 12:01.40 | dan_j | Compared to 11 |
| 12:01.53 | leifmadsen | depends what your testing tells you really |
| 12:02.03 | leifmadsen | I consider 12 a development preview |
| 12:02.14 | dan_j | a preview to 13? |
| 12:02.20 | leifmadsen | and actually mjordan said 12 is better because you can externalize more |
| 12:02.27 | leifmadsen | yea, 13 will be LTS |
| 12:03.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:04.22 | dan_j | Will there be major config changes between the last version of 12 and the new 13 when its released? |
| 12:05.32 | [TK]D-Fender | several=probably, major=few |
| 12:05.36 | [TK]D-Fender | (if any) |
| 12:08.04 | dan_j | The annoying thing is that I believe in 1.8 or earlier, i could record and transfer. And regardless of transfer, it appears that when i start recording in the dialplan and call more than 1 sip peer, it breaks recording. |
| 12:08.12 | dan_j | Trying to pinpoint the exact issue. |
| 12:08.37 | snmp | strange thing. iax2 show peers displays: master (null) (D) 255.255.255.255 0 (T) (E) OK (3 ms) |
| 12:09.17 | snmp | but in a second: master 192.168.0.100 (D) 255.255.255.255 4569 (T) (E) OK (4 ms) |
| 12:09.34 | snmp | suggestions? |
| 12:10.59 | snmp | peer is always qualified and reachable... but whereis ip.. |
| 12:12.16 | snmp | =/ |
| 12:12.31 | [TK]D-Fender | "in a second"? |
| 12:13.27 | snmp | yep interval is something about 5-10 secs |
| 12:14.22 | [TK]D-Fender | sounds more like reregistration timing |
| 12:14.27 | [TK]D-Fender | than straight qualify |
| 12:14.45 | snmp | thought of this |
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| 12:24.15 | Zogot | hmm, anyone have information about what asterisk does with the urn:uuid? |
| 12:24.34 | Zogot | as we think that that could be the problem, and its incorrectly using the urn:uuid of the wrong sip peer |
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| 12:59.18 | mirela666 | Hi is there any option for sip trunk as prefix ? to change the CALLERID(dnid) to ${prefix}${CALLERID(dnid)} ? |
| 12:59.34 | mirela666 | or it has to be changed in dialplan |
| 12:59.59 | WIMPy | Almost everything is done in the dialplan. |
| 13:00.18 | WIMPy | But I don;t think I really understand your question. |
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| 13:01.04 | mirela666 | well for example to fix outgoing /incomming dnid with prefix on the trunk settings :D |
| 13:02.03 | WIMPy | What "trunk settings"? Are you talking FreePBX or something? |
| 13:02.16 | mirela666 | no just plain sip.conf |
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| 13:02.57 | WIMPy | User contexts and your dialplan. |
| 13:02.59 | WIMPy | -r |
| 13:03.06 | mirela666 | if incoming dnid is 333555111, for example to set prefix=+33 and incoming call will land on context as +33333555111 |
| 13:03.27 | mirela666 | oki :D |
| 13:03.40 | WIMPy | Use another context and a Goto(). |
| 13:04.57 | mirela666 | hate that :) but will have to :) |
| 13:05.07 | mirela666 | thought might be a quick fix :D |
| 13:06.49 | [TK]D-Fender | [09:03]mirela666if incoming dnid is 333555111, for example to set prefix=+33 and incoming call will land on context as +33333555111 <- no. It arrives as it arrives. |
| 13:07.48 | mirela666 | indeed, thanks, logic starts on odialplan |
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| 15:44.52 | elbriga | Hi |
| 15:48.21 | elbriga | Could anyone point me where i can get info on the procedure that Asterisk does when trying to authenticate an incoming SIP? |
| 15:49.04 | elbriga | I have 2 * boxes trunked, but the extension calling the other box exists as a sip peer on it, its getting confusing.... |
| 15:51.25 | [TK]D-Fender | on each system set: |
| 15:51.30 | [TK]D-Fender | fromuser=THEUSERNAME |
| 15:51.37 | [TK]D-Fender | sendrpid=yes |
| 15:51.41 | [TK]D-Fender | trustrpid=yes |
| 15:51.49 | [TK]D-Fender | reload and all should be fine |
| 15:53.35 | elbriga | I'll test... but I think I tried and lost some CID info with this... be right back... |
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| 16:22.40 | ipengineer | Is there a way to log RTCP SR events to the CDR? If I enable rtcp I see the events being sent to the server from the user agent (if I am looking at it right) I would like to log the packet loss for the call to CDR. |
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| 17:10.27 | mjordan | ipengineer: yes. You can use the CDR function in combination with the CHANNEL function |
| 17:12.05 | mjordan | ipengineer: something like Set(CDR(rtcp_stat_remote_jitter)=${CHANNEL(rtpqos,audio,remote_jitter)}) |
| 17:12.47 | mjordan | ipengineer: you could do 'all', but since it will dump that out in comma delineated format, it would be highly dependent on your CDR backed on whether or not that is a "good idea" |
| 17:22.21 | *** join/#asterisk MATPOC (cf6b960a@gateway/web/freenode/ip.207.107.150.10) |
| 17:22.31 | MATPOC | Hi, I wanted to ask the Asterisk Community if there is any experience installing and using asterisk on cell phones? I put Ubuntu touch and asterisk on Nexus 5. SIP works great. How can I make asterisk talk to GSM module. Should one write custom channel driver? |
| 17:26.10 | WIMPy | AFAIK there is no API to communicate to the radio. |
| 17:26.52 | mjordan | But, assuming there was, then yes, you would write a channel driver |
| 17:27.00 | MATPOC | Ubuntu touch somehow makes and receives calls no problem. |
| 17:28.36 | MATPOC | thank you for your reply. I'll look into writing channel driver. I will take official docs, soruce and chan_dongle as an example |
| 17:29.10 | WIMPy | If they have an API, it's likely that you can just use chan_dongle. |
| 17:34.59 | MATPOC | good idea, I need to think about how to check that. Thanks for pointers. |
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| 17:35.58 | WIMPy | You will probably have to make sure the phone application doesn't run. |
| 17:37.01 | WIMPy | Hi newtonr. Missed you yesterday. |
| 17:37.12 | WIMPy | newtonr: Sorry. I was away. The issue I have is that chan_sip uses whatever you specify in the host=line as domain part of the to header. |
| 17:37.34 | WIMPy | So my question was if that can be solved with pjsip. |
| 17:38.16 | newtonr | WIMPy, ahh |
| 17:38.29 | MATPOC | WIMPy, thank you. I'll try brute force method - will just install dongle_chan and will monitor its communication |
| 17:39.08 | mjordan | MATPOC: btw, if you hit into specific coding questions with Asterisk, you may want to use #asterisk-dev |
| 17:40.21 | newtonr | WIMPy, i think using a dialstring like I mentioned should work, where you specify the domain. that should use what you set there instead of what is in the AOR/Contact |
| 17:40.31 | MATPOC | Jordan, thank you, I will |
| 17:43.47 | newtonr | WIMPy, http://tinyurl.com/n4ow23q the dialstring in that section with the explicit URI |
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| 18:19.14 | ipengineer | mjordan: Just walked back in Thanks! Do I need to have RTCP debug set to on via the CLI or is it already on? |
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| 18:28.53 | xaristax | Hi i hope someone can help me im trying to make a muted PAGE but i always get sound both sides how can i silence the other party |
| 18:30.23 | xaristax | im doing this http://pastebin.com/wwjYUkx8 |
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| 18:45.22 | WIMPy | newtonr: Ok, thanks. Now I just need to find some time to look in to the pjsip thing. |
| 18:45.55 | newtonr | WIMPy, if I can remember; in a bit I'll test it out to make sure it does what I am thinking. |
| 18:46.17 | WIMPy | cool |
| 18:46.55 | newtonr | at the moment, i'm busy writing out some device and extension state documentation for the wiki |
| 18:47.05 | WIMPy | Although I also seem to have an issue with IAX if Asterisk starts before DNS is working. |
| 18:47.25 | WIMPy | Documentation is a good thing. |
| 18:48.00 | newtonr | we are putting a lot of new documentation on the wiki :) |
| 18:48.25 | WIMPy | That's great to hear. |
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| 18:49.59 | newtonr | WIMPy, i think there is a IAX DNS issues on jira file recently.. was that you? i remember working on one earlier this week /me looks |
| 18:50.39 | WIMPy | No. My issue is rather strange, as usual. It contacts the right host, but on the wrong port. |
| 18:51.41 | WIMPy | Works after doing an 'iax2 reload'. Until then the port is shown as 0. But according to tcpdump it sends to 4620 IIRC. |
| 18:51.59 | newtonr | https://issues.asterisk.org/jira/browse/ASTERISK-23767 |
| 18:52.03 | WIMPy | Explicitely setting the port number in the peer entry doesn;t make a difference. |
| 18:52.16 | newtonr | the reporter does say "as it correclty switches the IP from 0.0.0.0 to whatever it should be but doesn't restore the port from 0 (when unavailable) to 4569 (when available). " |
| 18:52.48 | WIMPy | reads... |
| 18:53.22 | newtonr | i haven't had a chance to read his last comment. i just scanned it real quick. if that seems like your issue make sure to add any further debug or info to it |
| 18:53.32 | newtonr | goes back to documentation stuff |
| 18:53.56 | WIMPy | The initial description doesn;t fit. The IP is set correctly. But I'll read on. |
| 18:55.25 | WIMPy | Yes, 2nd comment is what it's about. |
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| 19:03.37 | mjordan | ipengineer: you should not need any rtp debugging enabled. |
| 19:08.34 | WIMPy | This is the one: ASTERISK-19106 is also true for IAX, and that still in 11.9. |
| 19:13.36 | ipengineer | mjordan: K thanks. |
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| 20:22.44 | pjensen00 | Hello! I'm using Asterisk 12. I'm trying to do a Page using the PJSIP channel driver. In 1.8 I was able to attach SIP headers to the outgoing leg using the SIPAddHeader command. However, the PJSIP command PJSIP_HEADER command attaches the header only to the channel that is using the command, not the channel that is spawned upon dialing. I was able to use the pre dial option on a 'Dial' command but there is no option to do that with |
| 20:22.45 | pjensen00 | <PROTECTED> |
| 20:22.45 | pjensen00 | I need to add a header to inform the endpoint to auto pick up and I can't figure out how to do this or if it's even possible. |
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| 20:33.57 | enoch | hi all, im trying channel_dongle. It works, but after the first call id doesn't work anymore and i have to restart asterisk to make it work again. |
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| 20:43.16 | rmudgett | pjensen00: You could page local channels which could then dial using a pre-dial handler. |
| 20:43.45 | pjensen00 | .... oh hey. That sounds like a really clever solution. |
| 20:43.53 | pjensen00 | thanks, I'll give that a shot. |
| 20:44.46 | mjordan | pjensen00: that is a good point on it being an oversight |
| 20:44.59 | mjordan | there's a few other things that don't have pre-dial either... |
| 20:45.19 | malcolmd | mjordan: app_page has pre-dial in trunk, no? |
| 20:45.38 | WIMPy | Like when you never dial? |
| 20:47.22 | rmudgett | The only applications that do pre-dial are Dial and FollowMe |
| 20:48.02 | WIMPy | A predialplan would be a lot more usefull than predial, IMHO. |
| 20:48.33 | rmudgett | Page and Queue could be made to do pre-dial if you were suitably ambishous... :) |
| 20:50.55 | rmudgett | s/ambishous/ambitious/ |
| 20:51.47 | mjordan | malcolmd: Hm. I don't think so... |
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| 20:52.01 | mjordan | and I see rmudgett beat me to it |
| 20:52.21 | marvinmartian | Is there a way to tell if a certain channel is causing asterisk to have high cpu utilization? |
| 20:53.15 | mjordan | lawl |
| 20:53.26 | mjordan | pjensen00: so, yeah, trunk has pre-dial for Page |
| 20:53.36 | pjensen00 | Hrm, since I haven't really taken a look into asterisk code much... how much work would it be to put the predial option in Page? :S |
| 20:53.49 | pjensen00 | oh.... in trunk? |
| 20:53.53 | mjordan | pjensen00: If you're okay with running a little bleeding edge... try trunk :-) It's not substantially different than 12 |
| 20:54.07 | pjensen00 | Interesting, I'll give it a look. Thanks! |
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| 20:57.17 | navaismo | ~ask |
| 20:57.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:05.03 | navaismo | ^*that was for me* |
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| 21:36.26 | *** join/#asterisk clembm (clembm@66.64.14.154) |
| 21:36.31 | clembm | Hello everyone |
| 21:37.10 | clembm | I have a stupid question. I have an asterisk server with a public interface and a private interface |
| 21:37.36 | clembm | Is there anything special I need to do? |
| 21:38.04 | WIMPy | Not unless the story continues. |
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| 21:40.10 | clembm | nope |
| 21:40.11 | clembm | haha |
| 21:40.16 | clembm | I have internal phones |
| 21:40.22 | clembm | and we have external users as well |
| 21:40.58 | carrar | Thats HOT++ |
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