similar to: pjsip module reload problem

Hi, On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START and BRIDGE_END events zone-s*CLI> module reload cel The module 'cel' reported a reload failure     -- Reloading module 'cel' (CEL Engine) [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 ast_cel_str_to_event_type: Unknown event name 'BRIDGE_START' [2020-07-10 17:57:01] ERROR[16163]:

On Wed, Jul 22, 2020 at 12:44 PM Administrator <admin at tootai.net> wrote: > No one on this ? > > Le 10/07/2020 à 18:06, Administrator a écrit : > > Hi, > > > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > > and BRIDGE_END events > > > > zone-s*CLI> module reload cel > > The module 'cel' reported a

Hi Austin, Alas, no, the program cannot be downloaded, but the CD can be ordered (free) from pariyatti.org. I would be happy to send one, but can only do so in a couple of weeks when I am in BKK (if you could send your mailing address to me privately, I would be more than happy to do so! ashin DOT sopaka at g(ee)mail dot c0m) More info about this can be found at http://www.tipitaka.org/ I also

Hi all, I've got an xts time series of stock symbols and closing prices. > head(x) symbol close 2010-01-04 "AFB" "13.46" 2010-01-04 "AKP" "12.80" 2010-01-04 "APX" " 8.78" 2010-01-04 "AYN" "13.15" 2010-01-04 "BAF" "13.50" 2010-01-04 "BBF" "12.86" >

[Accidentally sent to samba-technical --dave] Stubel Timo wrote: > Hello List, > > > > hope anybody can help me. We have problems with our squid porxy with userauthentication. > > > > The error is: > > authenticateNTLMHandleReply: Error validating user via NTLM. Error returned 'BH NT_STATUS_ACCESS_DENIED' > > > > We are using

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

-------- Original Message -------- Subject: smbstatus getting a SIGBUS error Date: Thu, 08 Jun 2000 13:52:13 +0700 From: Arnold Troeger <stssart@bkk.unocal.com> To: samba-bugs@samba.org CC: arnold.troeger@bkk.unocal.com I'm running Samba 2.0.7 and have been having a few problems with it. Just to keep things focused, I'll stick with smbstatus as this one has been the longest lasting

Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var:

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

Somemore information regarding the smbstatus error. The sigbus error is occuring where I noted below. I now have data values to go along with the location. The SIGBUS generating bit is "entry_scanner_p->e.pid" which looks to be in an unaccessable area of memory. Looking back through the code, I find a couple of bizarities. For example file_scanner_p->num_share_mode_entries

On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the

Is it possible to use serveral protocols for a single transport section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0.0.0 I get an error that transport-udp is not found. Do I need a dedicated interface for WebRTC? [Feb 15 12:42:10] ERROR[3308]:

Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of

I'm running asterisk 16 on Fedora 31. If I start asterisk as user asterisk, all goes well. But if I start asterisk from systemd: asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 sorcery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: config_options.c:710

Hi, I've run many time e2fsck, but in a special dir ls tells me: ls: r?cksendung-wlan.dvi: No such file or directory ls: baf?g_r?ckmeldung.latex: No such file or directory ls: finpr?f.pdf: No such file or directory $ cat finpr?f.pdf cat: finpr?f.pdf: Datei oder Verzeichnis nicht gefunden I don't know what to do? How can I find the failure? If I cat the files with debugfs, I see the

No one on this ? Le 10/07/2020 à 18:06, Administrator a écrit : > Hi, > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > and BRIDGE_END events > > zone-s*CLI> module reload cel > The module 'cel' reported a reload failure >     -- Reloading module 'cel' (CEL Engine) > [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 >

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Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, In Asterisk 11 it is possible to set extensions on DB table (sipppers) and also in sip.conf. But in Asterisk 13 apparently this is not possible: as I tried to set in ps_endpoints and also in pjsip.conf but only the realtime endpoints are loaded. Is there a way to use realtime + static modes at the same time for the ps_endpoints lookup using PJSIP. Thanks -------------- next part