similar to: SIP/2.0 489 Bad Event in reply to a PUBLISH

On Mon, Mar 23, 2020 at 7:15 AM John Hughes <john at calva.com> wrote: > Hi, in these dark days of COVID-19 lockdown I'm using linphone to > connect to my office asterisk system for working from home. > > It's going pretty well but the presence/BLF functions don't appear to work. > > In the linphone logs and asterisk debug I find that asterisk is > rejecting

So I've got a bit further with my  project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and

On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com> wrote: > So I've got a bit further with my project to get BLF working between > asterisk and linphone. > > Initially asterisk was rejecting linphone's SUBSCRIBE messages because > they didn't have an Accept: header. I've fixed that and now the initial > SUBSCRIBE messages work and I see all my

Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length): INVITE sips:bob at biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74bf9 Max-Forwards: 70 To: Bob <sips:bob at biloxi.example.com> From: Alice <sips:alice at

Credit goes to "cppcheck" Signed-off-by: Thomas Jarosch <thomas.jarosch at intra2net.com> --- common/common.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/common/common.c b/common/common.c index f443cb7..e8004d7 100644 --- a/common/common.c +++ b/common/common.c @@ -244,6 +244,7 @@ int sendsignalfn(const char *pidfn, int sig) if (fgets(buf,

On 23/03/2020 18:51, Joshua C. Colp wrote: > On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com > <mailto:john at calva.com>> wrote: > > > > Why is asterisk giving an error 500? I can find no reason, there > is nothing in any log. > > > The sequence number is from the past. The first SUBSCRIBE is sequence > number 22 (check the

> On 23/03/2020 18:51, Joshua C. Colp wrote: >> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com >> <mailto:john at calva.com>> wrote: >> >> >> Why is asterisk giving an error 500? I can find no reason, there >> is nothing in any log. >> >> >> The sequence number is from the past. The first SUBSCRIBE is

Asterisk can know that one of the attached phones is both "ringing" and "on the phone". However the sip NOTIFY it sends out to interested parties can only communicate one state, for example with pidf+xml it can either send "Ringing" or "On the phone" and so it sends "Ringing". This makes the "busy lights" less than useful, if a call

Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Best regards Le mer. 25 mars 2020 à 15:06, John Hughes <john at calva.com> a écrit : > > On 23/03/2020 18:51, Joshua C. Colp wrote: > > On Mon, Mar

Hi list, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? The trace looks like: ## PHONE -> ASTERISK ## PUBLISH sip:1001 at example.com SIP/2.0 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u;rport From: "1001" <sip:1001 at example.com>;tag=98slbhbn16 To: "1001" <sip:1001 at example.com> Call-ID:

On Wed, Jun 10, 2020 at 10:27 AM John Hughes <john at calva.com> wrote: > Asterisk can know that one of the attached phones is both "ringing" and > "on the phone". > > However the sip NOTIFY it sends out to interested parties can only > communicate one state, for example with pidf+xml it can either send > "Ringing" or "On the phone"

THis is the modified script that came with tinc 1.0pre2, and it now uses ifconfig in stead of ip-route. I changed the syntax to be correct (may already have been fixedin a newer version), it now checks whether or not there is a '/dev/tapX' or a '/dev/netlink/tapX' and it checks if there is a module or not. I also added the force_connect and the reload options. Force_connect send

Hi, I have an application which is dying horrible deaths (i.e. segmentation faults) in mid-flight, in production... And of course, I should fix it. But while I find and fix the bugs, I found something I think should be different - I can work on submitting a patch, as it is quite simple, but I might be losing something on my rationale. When Mongrel segfaults, it does not -obviously- get to clean

Hello, ? (using trixbox with Asterisk 1.6.0.26) ? I am looking for information about how Asterisk should notify the unavailable (SIP) state of a SIP device. ? I found out that the phone (SPA504G with attendant console) sends a SUBSCRIBE request with an Accept: application/dialog-info+xml. ? The situation is that the BLF leds are green even for phones that are currently not "online".

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.19.0 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:

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 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 have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost.