similar to: PJSIP Delay in Dialing

Sorry, should have included that. Asterisk 16.2.1 Mark. On Thu, 4 Apr 2019 at 14:56, Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote: > > As I understand it, delays like this are almost always caused by slow > > or failing DNS lookups. Running a packet capture on all interfaces > > filtering on port 53 shows no DNS

Thanks Joshua. Hopefully I'll be able to retry tomorrow. On Thu, 4 Apr 2019 at 15:30, Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Apr 4, 2019, at 11:27 AM, Mark Farmer wrote: > > Thanks, I did enable debugging but didn't see any attempts to resolve > > hostnames. I will give it another look. > > > > I did have an empty resolver_unbound.conf

Seems to be res_resolver_unbound.so Reading the documentation now but any hints greatly appreciated! Mark. On Thu, 4 Apr 2019 at 15:07, Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Apr 4, 2019, at 11:03 AM, Mark Farmer wrote: > > Sorry, should have included that. > > > > Asterisk 16.2.1 > > And what res_resolver module is loaded and in use? Depending

Thanks, I did enable debugging but didn't see any attempts to resolve hostnames. I will give it another look. I did have an empty resolver_unbound.conf (not even a general context) - would that likely cause issues? I would expect the defaults to kick in but I have now added: [general] hosts=system I will retest/debug when ASAP. Mark. On Thu, 4 Apr 2019 at 15:20, Joshua C. Colp <jcolp

On Fri, Apr 5, 2019, at 8:07 AM, Mark Farmer wrote: > Hi. Sadly I'm still struggling with this. I've captured the debug output. > I also upgraded to 16.3.0 this morning. > > https://pastebin.com/raw/HxbX0uXt This doesn't really give enough context to be able to really say anything. The SIP traffic, "pjsip set logger on", and normal verbose information mixed in

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

Hi everyone I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader & SIPRemoveHeader but the apps don’t appear to be installed in v16. Can anyone tell me where they went and how to get them installed please? Thanks Mark. Mark Farmer Senior UC Systems Architect Intercity Technology Limited HQ 101-114 Holloway Head, Birmingham, B1 1QP Tel: 0330 332 7933 / 07872542107 /

Hi all I am trying to use the cifs module to mount a share from a win2k3 server but I am getting the following error in /var/log/messages - the filesystem does mount but i'm having other issues accessing the data from our web based app. Can anyone explain why I get this error please? Indeed the file mount_cifs.so does not exist on the machine. automount[31080]: open_mount: (mount):cannot

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a ?crit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - --

Ludovic Gasc wrote: > 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>>: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > If CCSS is needed then the only option is to use chan_sip. The > > chan_pjsip module does not implement

Hello. I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip. Now i am transfering all from chan_sip to chan_pjsip. And have a lot of questions. First of... system: Asterisk 13.2 on slackware 14.1 Errors on outgoing call: [2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to create outgoing session to endpoint 'srv_d228' [2015-03-03 00:18:58]

Hi all, we are trying to move our servers from chan_sip to chan_pjsip. At this time no problems with phones, they all register fine and can place calls. But for a trunk we face problem and can't place calls despite the fact that registration is OK. What we get is: [2018-04-16 16:08:33] WARNING[18665]: res_pjsip_outbound_authenticator_digest.c:178

Ok so now I'm getting this when doing a make in asterisk... travis at pcimphone1:~/downloads/asterisk-13.5.0$ make [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC

Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named 'inband_progress' at line 867 of [May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317 sorcery_config_internal_load: Could not create an object of type

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE-----

Ok that did it after I did the steps to completely remove everything and do a new install. Thanks! > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 10:12 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

Hello! I'm trying to send a fax via T.38 to a destination, which should be T.38 capable. My provider supports T.38, too. Unfortunately, it doesn't work. This means: Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - for alaw again (and not for T.38)!! After about 30s, callee hangs up because of missing data (this is true, because I don't send alaw coded fax data.

Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the

Hi, after switching from chan_sip to chan_pjsip, a device running Grandstream Wave leads to the following error message on the asterisk console: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> <SSL routines- ssl3_get_client_hello-no shared cipher> len: 0 peer: 10.10.20.29:43357 Something with the encryption must have changed with asterisk. How can I get the device to