similar to: SIP/2.0 401 Unauthorized when calling from one SIP extension to another

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context | secret |

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so"

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Installed uploadify_rails3 gem, while submiting the form all params are correct but I am getting error "Completed 401 Unauthorized in 0ms" which prevents me from doing any further steps. If any of you has faced such situation please send me any solution to get rid of it. Thanks in advance -- You received this message because you are subscribed to the Google Groups "Ruby on

I integrated the my rails 2.x app with twitter. Im getting 401 issue after integrated the app with twitter. Can anyone say me what i missed?. -- Posted via http://www.ruby-forum.com/. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To unsubscribe from this group and stop receiving emails from it, send an email to

Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

Hello I use Asterisk 13 with FreePBX. When I try to connect my Softphone via VPN to Asterisk I'm registered and It's show via "pjsip list contacts" Then I try to call an internal number / other extension I get the following: "SIP/2.0 401 Unauthorized". The VPN net is list in pjsip.transports.conf:local_net=10.8.0.0/24 and

On 11/13/20 9:56 AM, PGNet Dev wrote: >> (2) see here: https://wiki.dovecot.org/Plugins/FTS/Solr >> >> two useful settings are debug and rawlog_dir=whatever to be added in the same line as fts_solr with fts_solr = ... debug line #35 @ https://pastebin.com/9ecLQspD _looks_ like the 401's from solr server itself. i do NOT currently have service indexer{} service

On 11/13/20 11:37 AM, John Fawcett wrote: >> still dunno why the 401. :-/ > > So I just did a quick check of running dovecot with a standalone > solr-8.7.0 instance and I'm not seeing any issues. +1 > I confirm I haven't configured anything for indexer or indexer-worker in > dovecot, just left the defaults. +1 > For 401's returned from your solr server

> I guess you didn't need to enclose username and password in quotes, i.e. > > fts_solr = > url=https://myuser:my%40pass at solr.example.com:8984/solr/dovecot/ > use_libfts soft_commit=yes batch_size=250 On 11/13/20 12:56 PM, John Fawcett wrote: > I guess you didn't need to enclose username and password in quotes, i.e. > > fts_solr = >

Hi, I have made a fresh install of asterisk-1.6.2.10 and when I register my soft phone it gives following error. Rest are default configurations. 32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER

On 13/11/2020 22:04, PGNet Dev wrote: >> I guess you didn't need to enclose username and password in quotes, i.e. >> >> fts_solr = >> url=https://myuser:my%40pass at solr.example.com:8984/solr/dovecot/ >> use_libfts soft_commit=yes batch_size=250 > > On 11/13/20 12:56 PM, John Fawcett wrote: > >> I guess you didn't need to enclose username and

Hi list! I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC behind NAT. From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the internet at 100.100.94.210 with a SIP account "3333" created in sip.conf: [3333] type=friend secret=something host=dynamic nat=yes qualify=no disallow=all allow=alaw allow=ulaw canreinvite=no context=voipin I dial +1234

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

On 13/11/2020 21:32, PGNet Dev wrote: > On 11/13/20 11:37 AM, John Fawcett wrote: >>> still dunno why the 401. :-/ >> >> So I just did a quick check of running dovecot with a standalone >> solr-8.7.0 instance and I'm not seeing any issues. > > +1 > >> I confirm I haven't configured anything for indexer or indexer-worker in >> dovecot, just

On 11/13/20 9:46 AM, John Fawcett wrote: >> (1) Can anyone yet verify Doveoct/fts-solr working with solr 8.7.x?>> (2) What logging config in Dovecot gets me more/detailed output for the fts_solr fail? >> (3) Any obvious clues as to what, specifically, is the source of this prob? >> > (1) I can't, I'm still on an earlier version I'll look for the 8.6.3,