[asterisk-users] Event showing who called who
I'm monitoring the ARI, and if extension 1 calls extension 2, it seems that extension 2 enters the bridge first, then extension 1 enters the bridge. Can I safely (always) determine who initiated the call by who is the latest endpoint to enter the bridge? Or is there a better way to know who initiates the call? I want to make this dialplan agnostic, so I don't want to listen for ChannelDialplan events. I see a Dial event that holds caller and peer information but that doesn't mean a bridge will be successful, so do I have to start tracking the id's of each endpoint, track who 'dialed', and once a bridge is entered try to related that data? Looking ahead I wonder how to handle if extension 1 calls a call group..but I'll ignore that for now. Thanks Brian (ast newb) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with pjsip
On Thu, Jun 8, 2023 at 9:41 AM Yves wrote: > 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: Error parsing from_user=75b55btqu...@orange-obs.fr at > line 0 of >== chan_pjsip.so => (PJSIP Channel Driver) > > 1) Error with "@" character which constitutes URI and authuser see > excerpt from pjsip.conf. > > [transport-udp] > type = transport > protocol=udp > bind=0.0.0.0:5060 > local_net=172.16.1.0/255.255.255.0 > > [reg_orange-obs.fr] > type = registration > retry_interval = 120 > max_retries = 10 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_orange-obs.fr > client_uri = sip:+3313445x...@orange-obs.fr > server_uri = sip:orange-obs.fr > > [auth_reg_orange-obs.fr] > type=auth > password=3314C9BA9688C2AA > username = 75b55btqu...@orange-obs.fr > > [Biv_Sortie] > type = aor > contact = sip:75b55btqu...@orange-obs.fr@orange-obs.fr > default_expiration = 3600 > > [Biv_Sortie] > type = identify > endpoint = Biv_Sortie > match = orange-obs.fr > > [Biv_Sortie] > type=auth > username = Biv_Sortie > password=3314C9BA9688C2AA > > [Biv_Sortie] > type=endpoint > context = Isdn_Inbound > dtmf_mode=rfc4733 > disallow=all > allow = g722, alaw, g729 > direct_media=no > trust_id_inbound = yes > send_rpid=yes > from_user = 75b55btqu...@orange-obs.fr > from_domain = orange-obs.fr > language = en > allow_subscribe = yes > auth = Biv_Exit > outbound_auth = Biv_Sortie > aors = Biv_Sortie > > Question how can I solve this character problem "@"? > The from_user is the username. It can't contain "@" or the domain. You've already set from_domain, so just set from_user to the username. > > 2) resolution of the orange-obs.fr DNS. I am attaching an extract from > the documentation that Orange issued in 2015 > > SIP/Internet is described in RFC3261 and following. THE > SIP/IMS is described by 3GPP standards. It's not the same > SIP. > In the Internet world, VoIP machines route > SIP messages to the IP addresses of the FQDNs of the SIP URIs > (VoIP domain). In the 3GPP world, SIP messages are > routed to an I/P-CSCF (depending on whether we are in interco or in > IPBX) which has a different FQDN from the VoIP domain. > > BIV SIP > > – P-CSCF FQDN: pcscfgm.orange-obs.fr, resolved by DNS > voice > – VoIP domain: orange-obs.fr, not resolved by voice DNS. ex : > INVITE sip:0142277...@orange-obs.fr SIP/2.0 > 2 > The VoIP/Internet machine will not be able to determine the address > recipient of SIP messages. > > run the command “nslookup pcscfgm.orange-obs.fr” and > note the returned IP address 217.167.210.X > – add this address in the /etc/hosts file of the PBX: > 217.167.210.X pcscfgm.orange-obs.fr orange-obs.fr You don't need to do /etc/hosts. Set an outbound_proxy on the endpoint and registration like so: outbound_proxy=sip:cscfgm.orange-obs.fr\;lr;\hide This will cause the SIP requests to get sent to "cscfgm.orange-obs.fr" but that won't appear in the SIP signaling. You'll probably have other issues that will require configuration changing, since providers using IMS infrastructure for general SIP always causes that. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with pjsip
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: Error parsing from_user=75b55btqu...@orange-obs.fr at
line 0 of
== chan_pjsip.so => (PJSIP Channel Driver)
1) Error with "@" character which constitutes URI and authuser see
excerpt from pjsip.conf.
[transport-udp]
type = transport
protocol=udp
bind=0.0.0.0:5060
local_net=172.16.1.0/255.255.255.0
[reg_orange-obs.fr]
type = registration
retry_interval = 120
max_retries = 10
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_orange-obs.fr
client_uri = sip:+3313445x...@orange-obs.fr
server_uri = sip:orange-obs.fr
[auth_reg_orange-obs.fr]
type=auth
password=3314C9BA9688C2AA
username = 75b55btqu...@orange-obs.fr
[Biv_Sortie]
type = aor
contact = sip:75b55btqu...@orange-obs.fr@orange-obs.fr
default_expiration = 3600
[Biv_Sortie]
type = identify
endpoint = Biv_Sortie
match = orange-obs.fr
[Biv_Sortie]
type=auth
username = Biv_Sortie
password=3314C9BA9688C2AA
[Biv_Sortie]
type=endpoint
context = Isdn_Inbound
dtmf_mode=rfc4733
disallow=all
allow = g722, alaw, g729
direct_media=no
trust_id_inbound = yes
send_rpid=yes
from_user = 75b55btqu...@orange-obs.fr
from_domain = orange-obs.fr
language = en
allow_subscribe = yes
auth = Biv_Exit
outbound_auth = Biv_Sortie
aors = Biv_Sortie
Question how can I solve this character problem "@"?
2) resolution of the orange-obs.fr DNS. I am attaching an extract from
the documentation that Orange issued in 2015
SIP/Internet is described in RFC3261 and following. THE
SIP/IMS is described by 3GPP standards. It's not the same
SIP.
In the Internet world, VoIP machines route
SIP messages to the IP addresses of the FQDNs of the SIP URIs
(VoIP domain). In the 3GPP world, SIP messages are
routed to an I/P-CSCF (depending on whether we are in interco or in
IPBX) which has a different FQDN from the VoIP domain.
BIV SIP
– P-CSCF FQDN: pcscfgm.orange-obs.fr, resolved by DNS
voice
– VoIP domain: orange-obs.fr, not resolved by voice DNS. ex :
INVITE sip:0142277...@orange-obs.fr SIP/2.0
2
The VoIP/Internet machine will not be able to determine the address
recipient of SIP messages.
run the command “nslookup pcscfgm.orange-obs.fr” and
note the returned IP address 217.167.210.X
– add this address in the /etc/hosts file of the PBX:
217.167.210.X pcscfgm.orange-obs.fr orange-obs.fr
Note that it works with sip.conf . The current installation is
operational with the information provided by /etc/hosts
below the debug in asterisk 19.6
[2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:115
custom_prepare: Skip: 0; SQL: SELECT * FROM ps_auths WHERE id = ?
[2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:134
custom_prepare: Parameter 1 ('id') = 'auth_reg_orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_odbc.c:808
ast_odbc_release_obj: Releasing ODBC handle 0x55855d1977d0 into pool
[2023-06-08 13:37:17] DEBUG[185433]: config.c:3847 ast_parse_arg:
extract uint from [32] in [0, 4294967295] gives [32](0)
[2023-06-08 13:37:17] DEBUG[185433]:
res_pjsip_outbound_registration.c:699 handle_client_registration:
Outbound REGISTER attempt 2 to 'sip:orange-obs.fr' with client
'sip:+3313445x...@orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:475
sip_resolve: Performing SIP DNS resolution of target 'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:502
sip_resolve: Transport type for target 'orange-obs.fr' is 'UDP transport'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:545
sip_resolve: [0x55855d769c88] Created resolution tracking for target
'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174
sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with
record type '35', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174
sip_resolve_add: [0x55855d769c88] Added target '_sip._udp.orange-obs.fr'
with record type '33', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174
sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with
record type '1', transport 'UDP transport', and port '5060'
[2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:616
sip_resolve: [0x55855d769c88] Starting initial resolution using parallel
queries for target 'orange-obs.fr'
[2023-06-08 13:37:17] DEBUG[185340]: dns.c:555 ast_search_dns_ex: DNS
search failed for orange-obs.fr
[2023-06-08 13:37:17] DEBUG[185340]: dns_system_resolver.c:154
dns_system_resolver_process_query: DNS search failed for query:
'orange-obs.fr'
[2023-06-08 13:37:17]
