similar to: pjsip insecure=port,invite

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

Right now it reply 401 Unauthorized with message in log "No matching endpoint ..." on Content 0 should reply 200 OK I guess <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> OPTIONS sip:10.30.100.27:5080 SIP/2.0 Via: SIP/2.0/UDP 10.30.100.41;branch=z9hG4bKf5eb.1ac76487000000000000000000000000.0 To: <sip:10.30.100.27:5080> From: <sip:vprx00 at

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

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

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_username at sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] type=endpoint context=from-my_provider

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Wiky needs to be updated https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29 This is the example shown: "[6001] endpoint=6001 match=203.0.113.1" It should be: "[6001] type=identify endpoint=6001 match=203.0.113.1"

09.09.2016 13:45, Joshua Colp ?????: > Dmitry Melekhov wrote: >> Hello! >> >> >> Upgraded 13.10 to 13.11.1 today and now I see messages in log: >> >> >> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request >> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for >>

14.09.2018 15:43, Jiri Denemark пишет: > On Thu, Sep 13, 2018 at 19:37:00 +0400, Dmitry Melekhov wrote: >> >> 13.09.2018 18:57, Jiri Denemark пишет: >>> On Thu, Sep 13, 2018 at 18:38:57 +0400, Dmitry Melekhov wrote: >>>> 13.09.2018 17:47, Jiri Denemark пишет: >>>>> On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote:

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP tcp on port 5068? telnet localhost 5068 Trying 127.0.0.1... telnet: connect to address 127.0.0.1:

13.09.2018 18:57, Jiri Denemark пишет: > On Thu, Sep 13, 2018 at 18:38:57 +0400, Dmitry Melekhov wrote: >> >> 13.09.2018 17:47, Jiri Denemark пишет: >>> On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote: >>>> After some mistakes yesterday we ( me and my colleague ) think that it >>>> will be wise for libvirt to check config file existence

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

13.09.2018 17:47, Jiri Denemark пишет: > On Thu, Sep 13, 2018 at 10:35:09 +0400, Dmitry Melekhov wrote: >> After some mistakes yesterday we ( me and my colleague ) think that it >> will be wise for libvirt to check config file existence on remote side > Which config file? > VM config file, namely qemu. We forgot to mount shared storage (namely gluster volume), on which we