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| 02:15.19 | cajalcrusher | Does anyone know of a stable softphone for Linux that supports TLS, SRTP and IPv6? I normally use Blink but it doesn't support IPv6. The new version of Linphone on Linux is very unstable. |
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| 16:18.18 | maximCH | ... I'm moving from chan_sip to chan_pjsip ... in the dialplan I used to use Dial(SIP/mytrunk/${EXTEN}) ... I guess that is no longer supported and you need to use Dial(PJSIP/${EXTEN}@mytrunk) ... :( |
| 16:24.00 | Samot | Right. |
| 16:24.06 | Samot | You can dial PJSIP endpoints differently |
| 16:24.14 | Samot | Because PJSIP has different things. |
| 16:24.58 | maximCH | yah ... it just means I have to change my dialplan as well |
| 16:25.13 | Samot | Yes |
| 16:25.22 | maximCH | and I've been using variables like TRUNK=SIP/mytrunk/00 .... to quickly change my trunks |
| 16:25.27 | Samot | Because functions for Chan_SIP don't work for Chan_PJSIP. |
| 16:25.31 | maximCH | yup |
| 16:25.34 | maximCH | oh well |
| 16:25.37 | Samot | SIPHEADER doesn't work for Chan_PJSIP |
| 16:26.13 | maximCH | yeah ... I haven't figured out yet how to extract P-Asserted-Id from the headers ... but that's not that important at the moment. |
| 16:26.59 | maximCH | I like the pjsip wizard though and the templates... that saved a lot of time. |
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| 17:05.20 | maximCH | ok .. but I don't get how to dial a SIP URI with PJSIP without using a predefined trunk |
| 17:27.25 | maximCH | I guess you just need to add an arbitrary trunk ... |
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| 20:33.45 | grummund | is a bit confused... |
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| 20:34.17 | grummund | sample sip.conf says "Don't mix extensions with the names of the devices. Devices need a unique name. The device name is *not* used as phone numbers." |
| 20:35.20 | grummund | ok, i get this... but how is callerid set up when one extension calls another? |
| 20:37.04 | grummund | at the moment when calling between extensions callerid shows the caller device name. |
| 20:39.33 | grummund | also 'sip show peers' lists each Name/username as an identical pair blah/blah for each device. |
| 21:17.50 | drmessano | grummund: I would suggest using pjsip instead |
| 21:17.55 | drmessano | chan_sip is kinda dead |
| 21:19.33 | grummund | drmessano: so it seems but the (excellent imho) book at asteriskdocs.org is all about chan_sip. |
| 21:20.50 | grummund | looks like i may have it fixed though using 'callerid=...' in sip.conf |
| 21:21.12 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Home |
| 21:21.16 | grummund | pjsip is on the todo list to play with later. |
| 21:22.14 | grummund | thanks :-) so much to learn... |
| 21:22.28 | drmessano | No point wasting time learning chan_sip |
| 21:28.36 | grummund | what does the pj stand for? |
| 21:31.13 | drmessano | It's from the project founders last name |
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| 22:19.27 | sibiria | is there a way to make asterisk either reject invitations, or just not send rtp at all, for cases where the invitation comes from an external address but the SDP describes a private network as the origin? |
| 22:20.38 | sibiria | origin and/or connection data fields |
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| 23:15.22 | sibiria | or otherwise in some way force rtp to be returned to the external address the invitation came from, in that case |
| 23:25.20 | Samot | Symmetrical RTP |
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| 23:28.56 | sibiria | hm that's something else, to make sure RTP is sent and received from the same single port instead of using one to receive and another to send from. what i want to avoid are cases where an external entity dials in and says "send rtp back to 192.168.1.4" which obviously causes asterisk to hurl RTP into the boid - or worst case at some computer in its LAN |
| 23:30.09 | sibiria | void* |
| 23:35.26 | Samot | Use Kamailio |
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