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| 00:18.05 | ak77 | hello all |
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| 02:47.10 | jpadawan | Hi. I using a simple php page with AMI to perform a click2call. Is possible to send an id or variable and store it on cdr? i want to associate the call with an external crm page. |
| 02:47.21 | jpadawan | $ami->action({ 'Action' => 'Originate'....}) |
| 02:53.28 | [TK]D-Fender | yes |
| 02:53.37 | [TK]D-Fender | Go read the instructings for the AMI action |
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| 03:58.35 | sparrowvivek16 | wcfxo: Out of space to write register 05 with 0a |
| 03:58.45 | sparrowvivek16 | Asterisk with x100P |
| 03:58.51 | sparrowvivek16 | IRQ is too low... |
| 03:58.57 | sparrowvivek16 | how do i correct it? |
| 04:01.14 | sparrowvivek16 | open suggestions |
| 04:03.12 | sparrowvivek16 | guys please help me... |
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| 04:26.19 | talntid2 | I installed 13.6 from source. how do you guys generally handle starting it as a service on ubuntu 13? |
| 04:26.22 | talntid2 | i mean buuntu 14 |
| 04:27.07 | talntid2 | i have used packages in the past, but wanted the newer pjsip stuff, so i went from source... now it doesn't start automagically, do you just use a @reboot cronjob, or make it a service/ |
| 04:27.56 | talntid2 | seems like there should be init scripts somewhere |
| 04:28.45 | talntid2 | nevermind. make config |
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| 05:36.40 | dovid | so who here is @ Astricon? |
| 05:44.29 | talntid2 | not me |
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| 05:47.40 | talntid2 | I'm attempting to use pjsip + realtime config to register to my ITSP. |
| 05:48.01 | talntid2 | I have the AOR, Auth, and Endpoint, but when I attempt registration, it doesn't seem to "take" |
| 05:49.30 | talntid2 | I used Alembic to do the database schema, but I don't see anything in pjsip.conf pointing at that table (ps_registrations) |
| 05:49.32 | talntid2 | any ideas? |
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| 09:24.41 | wasanzy | hello |
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| 09:26.09 | wasanzy | when I send OBD, the url is like this: from '<sip:33301@IP>;tag=as0ccc85f9' but instead of tag, I wan user=phone. I added this to the sip config usereqphone = yes but it didn't change anything |
| 09:26.14 | wasanzy | how else can I do this? |
| 09:30.44 | ak77_ | I have asterisk and pjsip setup with outbound registration over tls. Registration works, calls don't... I find strange that To: field has to have "sips:" as scheme in URI, otherwise Asterisk will just fails with hangup cause 34 (No circuit/channel available), never issuing an INVITE. |
| 09:32.05 | wasanzy | am using asterisk version 11.31 |
| 09:41.25 | ak77_ | my ITSP says Not supported URI on To: sips:..... |
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| 10:46.30 | ak77_ | maybe this has something to do with.... outbound_proxy should be set to sips:SERVERADDR, server URI then uses only sip:, and To: header as well |
| 10:47.54 | ak77_ | but now... asterisk is unwilling to REGISTER if transport is set to tls, and in wizard, server_uri_pattern=sips:${REMOTE_HOST} is not present. |
| 10:48.14 | ak77_ | why wouldn't it connect/register using proxy to sip:REMOTE_HOST ? |
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| 12:11.44 | wasanzy | hello any help? |
| 12:12.21 | wasanzy | I want the from url to have user=phone but it is not working for me that way. the to field has it |
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| 14:07.26 | Reapster | heya, I'm trying to get sorcery working with pjsip, using freepbx distro, all odbc and mysql stuff seems ok, did the alembic migrations to create db tables etc...but queries seem to fail when trying to update ps_endpoints table |
| 14:07.46 | Reapster | the query seems to have like 80 paramaters, is it possible its being truncated to 255 chars somewhere and failing? |
| 14:08.45 | Reapster | eg: DEBUG[7932]: res_config_odbc.c:113 custom_prepare: Skip: 32; SQL: UPDATE ps_endpoints SET timers_sess_expires=?, device_state_busy_at=?, dtls_cipher=?, from_domain=?, dtls_rekey=?, direct_media_method=?, send_rpid=?, pickup_group=?, sdp_session=?, dtls_verify=?, message_context=?, mailboxes=?, named_pickup_group=?, reco not sure if the logging is truncating the query, but odbc trace also seems to think its only 255 chars |
| 14:09.01 | Reapster | (* 13.5) |
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| 14:23.24 | Cuzner | is there a live stream of the devcon going on? |
| 14:31.35 | newtonr | Cuzner, yah, let me find the link |
| 14:31.47 | newtonr | https://wiki.asterisk.org/wiki/display/AST/AstriDevCon#AstriDevCon-AudioConference |
| 14:33.43 | Cuzner | danke |
| 14:36.04 | malcolmd | join #astridevcon to follow along |
| 14:37.09 | Cuzner | the audio is too choppy to follow along without wanting to flip a table. |
| 14:37.46 | Cuzner | 3rd world floridian internets :P |
| 14:38.14 | Reapster | heh come to south africa and see 3rd world internets ;) |
| 14:38.30 | malcolmd | periscope of devcon - https://t.co/RwlCHJAHo3 |
| 14:38.45 | malcolmd | we're also going to try to work on the normal audio stream during the break we're now taking |
| 14:39.44 | Cuzner | kk, yeah, that periscope looked/sounded fine... no lag (while it was up) |
| 14:40.14 | Cuzner | but the audio stream is choppy as hell, especially with volume changes. |
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| 15:55.59 | Janos | hey there, got a small problem with asterisk calendar, if I change an event it does not pick it up, is there any cache involved in this ?, is there a way to ask for a complete calendar refresh ? this is asterisk 11 |
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| 16:11.12 | *** join/#asterisk CFX_ (~deserver@unaffiliated/cfx-/x-0311648) |
| 16:11.53 | CFX_ | Is there a way to pickup a specific extension that's ringing? I know that you can use *8 to pickup a call coming in that's in a specific call group. |
| 16:14.25 | [TK]D-Fender | "core show applications like pickup" <- |
| 16:15.55 | CFX_ | It shows Directed call pickup but by default this is not assigned to a softkey? |
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| 16:26.57 | [TK]D-Fender | What is "it"? |
| 16:27.06 | [TK]D-Fender | And we have no idea what else has been configured.... |
| 16:29.50 | [TK]D-Fender | BRB |
| 16:31.35 | CFX_ | Pickup: Directed extension call pickup. |
| 16:31.35 | CFX_ | PickupChan: Pickup a ringing channel. |
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| 16:39.19 | robmal | CFX_: You need to set up something like exten => _123XXX,1,Pickup(${EXTEN:3}@internal) |
| 16:40.00 | CFX_ | ok thank you |
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| 17:23.51 | Leseratte | Hi, I have a little problem with my Asterisk server: My provider sometimes replies to my REGISTER with a 403 even when my password is perfectly valid. The solution is simple: Stop asterisk and start it again to re-register and everything works. However, I don't receive incoming calls while I'm not registered and Asterisk doesn't seem to try to register again on its own - "sip show registry" showed me that the "Reg. Time" was about 12:40 pm and i |
| 17:23.51 | Leseratte | t still didn't re-register until 7:20 pm when i restarted asterisk ... |
| 17:24.04 | Leseratte | Can I somehow force asterisk to re-register on errors? |
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| 19:20.43 | Dick-Tracy | Afternoon #asterisk -- Quick question for you guys, Is it possable to set VM operator per voicemail group, and not server wide |
| 19:20.57 | Dick-Tracy | i've looked online and havent seen anything |
| 19:21.01 | robmal | Ye. |
| 19:21.51 | WIMPy | Dick-Tracy: Use contexts |
| 19:22.21 | [TK]D-Fender | Operator depends on where you are in your dialplan |
| 19:22.44 | WIMPy | Or make a descision in the operator extension. |
| 19:23.11 | Dick-Tracy | so append it to the end, like hide from dir |
| 19:23.39 | WIMPy | ? |
| 19:24.24 | igcewieling | Dick-Tracy: pressing 0 (zero) simply jumps to extension o (oh). The VM option enables or disables that. |
| 19:24.40 | igcewieling | where exten => o goes to depends on the context. |
| 19:24.57 | Dick-Tracy | its the vm option that im trying to enable |
| 19:25.03 | Dick-Tracy | but not globally |
| 19:25.38 | igcewieling | Dick-Tracy: does the sample config file show anything useful? |
| 19:25.47 | WIMPy | IIRC it only works if the extension actually exists. So back to contexts. |
| 19:26.32 | igcewieling | Dick-Tracy: does it work if you enable globally? |
| 19:26.50 | Dick-Tracy | it does if enabled globally |
| 19:28.54 | igcewieling | Dick-Tracy: it shows you how to enable per mailbox in voicemail.conf.sample |
| 19:30.45 | igcewieling | line 415 on the version I have |
| 19:31.20 | Dick-Tracy | Thanks igcewieling, as i thought appended to the end |
| 19:31.48 | Dick-Tracy | doesnt work though, only when i got golbal turned on |
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| 19:32.14 | Dick-Tracy | -- i'll mess with it more, see what i can do |
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| 19:59.09 | talntid2 | I'm attempting to use pjsip + realtime config to register to my ITSP. |
| 19:59.10 | talntid2 | I have the AOR, Auth, and Endpoint, but when I attempt registration, it doesn't seem to "take" |
| 19:59.13 | talntid2 | I used Alembic to do the database schema, but I don't see anything in pjsip.conf pointing at that table (ps_registrations) |
| 19:59.39 | talntid2 | any ideas? Been playing with it, but don't "get it" yet, or I have something else going on and don't understand it enough to realize that |
| 20:00.04 | talntid2 | I have 2 SIP phones registered to the system, and I feel like it should be exactly the same, except a registration, but that doesn't seem to be working for me |
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| 20:36.23 | Dick-Tracy | AH HA apprently there was a whitespace before that VM users context, dont know why that caused only operatator to break, but never the less it works now, thanks WIMPy, robmal & igcewieling |
| 20:38.30 | robmal | Good job. |
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| 20:41.06 | stephan48 | can i trigger a callflow when somone just opens the line and does not dial anything for a few seconds? |
| 20:41.17 | stephan48 | what would i need to configure in extensions.conf for that? |
| 20:45.34 | WIMPy | [21:22] darkbasic has left IRC (Ping timeout: 240 seconds) |
| 20:45.40 | WIMPy | [21:24] ed_ has joined #asterisk (~zed_@82.179.199.146.dyn.plus.net) |
| 20:45.44 | WIMPy | Argh |
| 20:45.58 | WIMPy | stephan48: What kind of phone? |
| 20:46.29 | stephan48 | random softphones.. someone who is connected via isdn and the fritzbox sip client |
| 20:47.18 | WIMPy | If the phones are not physically connected to some hardware in your Asterisk box: no. |
| 20:47.32 | WIMPy | You have to configure that on the phone. |
| 20:47.46 | WIMPy | SIP doesn't "open" a line until you have finished dialling. |
| 20:48.03 | WIMPy | Don't know if the FB has such a feature. |
| 20:48.23 | stephan48 | they got some sip server/client thingy |
| 20:48.39 | stephan48 | you can connect dect and isdn phones to the client |
| 20:48.39 | WIMPy | With a real phone on the FB you can do it if you connect it to your Asterisk via remote CAPI. |
| 20:48.45 | stephan48 | ah okey |
| 20:49.06 | stephan48 | sounds like too much effort for a convinience feature |
| 20:49.58 | stephan48 | my idea was to present informations to the user.. like a menu when he does not dial a specific extension or outgoing number |
| 20:50.20 | stephan48 | will just put that on some extension |
| 20:50.24 | stephan48 | thanks throu! |
| 20:50.27 | WIMPy | Usully impossible with SIP. |
| 20:51.24 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 20:52.32 | WIMPy | Off course you could connect the FB to an ISDN card in your server. Then it would at least work with real phones on it. |
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| 20:53.31 | stephan48 | mh. i don't think hetzner would provide me with a isdn line into my root and then i would have to find someone who drags a few ISDN cords nearly throu all around germany |
| 20:53.55 | stephan48 | also i read that ISDN with asterisk is not that much un |
| 20:54.37 | WIMPy | Well, it works ok. But you have to make choices. |
| 20:54.43 | stephan48 | heh yea |
| 20:54.52 | stephan48 | i decided that i don't want to bother |
| 20:55.08 | WIMPy | has to |
| 20:55.12 | stephan48 | heh |
| 20:55.37 | stephan48 | using asterisk as sip server.. after i ditched realtime it works quite nicely with a few sip clients and 2 external providers |
| 20:56.45 | stephan48 | still need to find something that works on my android throu.. sipdroid allows me to accept calls but does not SRTP/TLS... zoiper.. allows tls/srtp, sees incoming calls but declines them directly... csipsimple has srtp/tls but does not show incoming calls.. |
| 20:57.46 | WIMPy | Yeah, I guess that's how SIP "works quite nicely" ... |
| 20:58.14 | stephan48 | i think i will settle on sipdroid and my always on vpn tunnel |
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| 21:15.07 | stephan48 | and the zoiper problem turned out to me a missing encryption=yes in the sip channel |
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| 21:27.30 | bkruse | Hello my Asterisk friends! I have a question about outbound failover. I have Asterisk 13.5.0 and I am looking to do a failover based on a SIP response. Is this possible in the dialplan now? |
| 21:27.58 | bkruse | I have an SBC that controls all outbound dialing, but in the case of an absolute SBC primary and secondary failure (like a timeout or 503), I'd like to failover to an emergency SIP route |
| 21:28.31 | WIMPy | Sure |
| 21:30.08 | bkruse | Ahh, looks like it was introduced in 1.8 - SIP_CAUSE - nice |
| 21:31.13 | WIMPy | And functions like HANGUPCAUSE. |
| 21:39.49 | *** join/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 21:52.27 | talntid2 | is using pjsip worth it? i am having a ton of trouble with getting stuff to work.. |
| 21:56.09 | WIMPy | Do you want the new features? |
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| 22:14.45 | talntid2 | I'm not sure. I like the orginization structure better, but I guess I don't properly understand it because I can't really get it to work all the way. maybe it is mixing it with realtime that is causing my issues though, i am not sure |
| 22:16.51 | rmudgett | pjsip uses sorcery.conf to determine where to get the config options |
| 22:18.10 | talntid2 | yeah, I have realtime working properly it seems... I can register sip phones to the system... |
| 22:18.32 | talntid2 | but like, if I dial to an extension, asterisk shows it is playing tt-weasels (my test extension), but I don't hear it on the sip phone |
| 22:18.54 | talntid2 | normally I'd think firewall, but the pbx is not firewalled at all, and not NAT'd.. |
| 22:19.06 | talntid2 | and the sip phone also can connect directly |
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| 22:26.41 | talntid2 | any advice on troubleshooting that? |
| 22:27.55 | robmal | No dialplan paste = no help here |
| 22:28.31 | talntid2 | robmal, this isn't a dialplan issue |
| 22:28.49 | robmal | Sure it can be. |
| 22:30.25 | talntid2 | it is getting to the right part of the dialplan... -- <PJSIP/101-00000012> Playing 'tt-monkeys.gsm' (language 'en') |
| 22:30.46 | talntid2 | just, can't hear that via the sip phone, which wouldn't be a dialplan issue |
| 22:31.36 | robmal | K, share the sip.conf and what happens during the call. |
| 22:32.24 | talntid2 | http://pastebin.com/HKLq6SMW |
| 22:32.54 | talntid2 | I'm using PJSip, so the sip.conf shouldn't really be doing anything if I am correct |
| 22:33.02 | talntid2 | i even unload the module chan_sip |
| 22:34.14 | talntid2 | here's the pjsip settings for the endpoint: http://puu.sh/kJiFa/f210412c33.png |
| 22:34.46 | robmal | Meh, if you want help you should be providing anything that's related... Have you seen recent dolphin photos? |
| 22:35.03 | talntid2 | http://puu.sh/kJiIB/8147c270d5.png |
| 22:35.33 | talntid2 | well I'm down to provide anything related, but I don't know that sip.conf is related, so I don't know if I should supply it |
| 22:35.52 | talntid2 | same thing like... moh.conf, I assume that isn't related, so I don't want to provide it and clutter things up, y'know? |
| 22:36.31 | robmal | All this looks like some other providers PBX which name if you mention will lead you to 'go to #freepbx' |
| 22:37.04 | talntid2 | negative, this is Asterisk 13.6, with Realtime Conf, and PJSip |
| 22:37.22 | talntid2 | not freepbx |
| 22:37.32 | robmal | I havent |
| 22:37.48 | robmal | seen any asterisk gui for some time |
| 22:37.53 | talntid2 | 13.6 from source, realtime from the asterisk docs (realtime works), and PJSip from the asterisk docs |
| 22:37.58 | talntid2 | I am not using any asterisk GUI |
| 22:38.15 | talntid2 | the screenshot I sent you is from a database, not an asterisk GUI |
| 22:38.37 | talntid2 | realtime configs are stored in postgres database, and the screenshot i sent you just shows the database rows |
| 22:38.38 | robmal | K, nevermind, too lazy to check ;-) |
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| 22:41.45 | robmal | <EOL> |
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| 23:09.15 | spicyramen_ | hi, i alwasy do asterisk -r, then core restart now, and then asterisk -r is there a way to do all that in one command? |
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| 23:43.27 | WIMPy | spicyramen_: asterisk -x |
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