similar to: set codec based on B side

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote: > The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk I just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several

Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. That is why I will post my questions here: 1- Transcoding: is this when

Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function "sip_write" inside "chan_sip.c". In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

On Fri, Jun 5, 2020 at 6:02 AM Marek Greško <mgresko8 at gmail.com> wrote: > Hello, > > I would like to ask about current state of subscribecontext in pjsip. > I found out some 6 years old discussion on that without any plans to > implement it in the future. > > I have phones in different contexts. I suspect, when I use its context > to subscribe, they will not see

Hi, Greetings of the day, Our configuration is like: We have installed both glusterFS server and GlusterFS client on node1 as well as node2. We have mounted node1 volume to both nodes. Our use case is : >From glusterFS node 1, we have to take an exclusive lock and open a file (which is a shared file between both the nodes) and we should write/read in that file. >From glusterFS node 2, we

Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels

Hello, On one hand, sorry to hear that higher polling frequency in upsmon did not help. On another, question is if the driver gets the info (online state and its changes) from device quickly enough. Initially I meant for you to also try if the "pollonly" flag (in each device section of ups.conf for usbhid-ups instances) would make a difference?.. I'd expect data-transfer interrupts

Hello, I'm looking for advices on IMAP config tuning for best user experience with Outlook 365. I'm currently using dovecot 2.3.4.1 (f79e8e7e4) provided with Debian 10. One of my users has Outlook 365 and an IMAP mailbox of large size with several folders (more than 3GB). From time to time as his main inbox folder is growing he can see some problems with its Outlook application.

Hello, Yes, "pollonly" is a driver option for certain devices (and relevant to just some drivers). Disconnects are probably relevant, at least to loss of connection (and staying that way, with less agressive retries in NUT v2.7.4 and before). There is a logged issue that "pollonly" mode might have trouble detecting a disconnection/staleness, I'm not sure there's merit

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between