Re: [OpenSIPS-Users] opensips call center module

to state Terminated sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time) | sip.Invitation | Session B2B.245.4610557.1715138504.19934727795b311b272f7dc2a4783f192166366d5a-4bdb in state Terminated is being disposed On Tue, 7 May 2024 at 17:20, Prathibha B wrote: > Getting bl

Re: [OpenSIPS-Users] opensips call center module

] >> >> The agent state shows status as incall. >> >> On Tue, 7 May 2024 at 06:35, Jehanzaib Younis >> wrote: >> >>> Hi Parathiba, >>> >>> Could you capture the SIP packets?

Re: [OpenSIPS-Users] opensips call center module

After about 45 seconds, the screen on the receiver side becomes blank. On Tue, 7 May 2024 at 12:32, Prathibha B wrote: > I delete the records from cc_calls table and restart opensips. Then the > call is getting is transferred to one of the agents. What is the solution > to this?

Re: [OpenSIPS-Users] opensips call center module

to the same agent. The call is not going to the next available agent. On Tue, 7 May 2024 at 11:32, Prathibha B wrote: > "Agents": [ > { > "id": "101001", > "Ref": 1, >

Re: [OpenSIPS-Users] opensips call center module

ot;Ref": 1, "Loged in": "YES", "State": "incall" } ] The agent state shows status as incall. On Tue, 7 May 2024 at 06:35, Jehanzaib Younis wrote: > Hi Parathiba, > > Could you capture the SIP packet

[OpenSIPS-Users] opensips call center module

I'm able to hear the message given in the message queue but the call is not getting transferred to the online agent, -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Send failed

sip trace attached On Fri, 3 May 2024 at 06:01, Prathibha B wrote: > # TCP Connection settings > tcp_connect_timeout = 200 > tcp_connection_lifetime = 3600 > tcp_max_connections = 4096 > tcp_keepalive = 0 > tcp_keepcount = 5 > tcp_keepidle = 60 > tcp_keepinterval =

Re: [OpenSIPS-Users] Send failed

# TCP Connection settings tcp_connect_timeout = 200 tcp_connection_lifetime = 3600 tcp_max_connections = 4096 tcp_keepalive = 0 tcp_keepcount = 5 tcp_keepidle = 60 tcp_keepinterval = 0 On Fri, 3 May 2024 at 05:45, Prathibha B wrote: > How to check TCP keepalive from client and server? >

Re: [OpenSIPS-Users] Send failed

ttings are correct and are aligned with what the far end expects? Have you verified the message is being sent to the correct remote target to begin with? Ben Newlin From: Users mailto:users-boun...@lists.opensips.org>> on behalf of Prathibha B mailto:prathibhab@gmail.com>> Dat

Re: [OpenSIPS-Users] Send failed

Can someone point me on how to resolve this issue? On Wed, 1 May 2024 at 18:49, Prathibha B wrote: > How to solve this error- 477 send failed. > -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.o

[OpenSIPS-Users] Send failed

How to solve this error- 477 send failed. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Call Center error

If 1 agent rejects the call, will the call be forwarded to the next logged in agent? On Mon, 29 Apr 2024 at 14:30, Prathibha B wrote: > The call lands in agent2. But the call stops after 5 seconds in agent2. > The MOH continues in the call initiator. > > On Wed, 24 Apr 2

Re: [OpenSIPS-Users] Call Center error

The call lands in agent2. But the call stops after 5 seconds in agent2. The MOH continues in the call initiator. On Wed, 24 Apr 2024 at 20:05, Prathibha B wrote: > config file attached. > > On Wed, 24 Apr 2024 at 19:51, Prathibha B > wrote: > >> Now I've migrated

[OpenSIPS-Users] opensips

Does opensips call center module work for webrtc based video calls? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Call Center error

o dialog found reply 200 for method INVITE, On Sat, 20 Apr 2024 at 20:51, Prathibha B wrote: > when I run > > opensips-cli -x mi cc_list_calls > { > "Calls": [ > { > "id": "219.3", > "Ref": 2,

Re: [OpenSIPS-Users] call not landing in agents

Now the call is landing to an agents. But the music on hold continues without stopping. How to resolve this? On Mon, 22 Apr 2024 at 17:24, Prathibha B wrote: > I've setup a call center using opensips. When I call to a number, the call > should land in one of the call center agents. I've b

[OpenSIPS-Users] call not landing in agents

I've setup a call center using opensips. When I call to a number, the call should land in one of the call center agents. I've browser phone as agent1 and zoiper phone as agent2. The call is landing in agent2 after 45-50 mins of call establishment. But the call never lands in browser phone. --

Re: [OpenSIPS-Users] Call Center error

ot;Loged in": "YES", "State": "wrapup", "Wrapup-ends": 16 }, { "id": "101002", "Ref": 0, "Loged in": "YES", "State&

Re: [OpenSIPS-Users] Call Center error

A non stop music is getting played but the call is not getting redirected to the call center agent. On Sat, 20 Apr 2024 at 14:17, Prathibha B wrote: > I am getting 401 Unauthorized error. How to resolve this issue? > > On Fri, 19 Apr 2024 at 19:13, Prathibha B > wrote: > >>

Re: [OpenSIPS-Users] Call Center error

I am getting 401 Unauthorized error. How to resolve this issue? On Fri, 19 Apr 2024 at 19:13, Prathibha B wrote: > Ok. > > Sent from Outlook for Android <https://aka.ms/AAb9ysg> > -- > *From:* Users on behalf of Brett > Nemeroff > *Sent

Re: [OpenSIPS-Users] Call Center error

:09 AM Prathibha B mailto:prathibhab@gmail.com>> wrote: or freeswitch as the media server. On Fri, 19 Apr 2024 at 14:45, Prathibha B mailto:prathibhab@gmail.com>> wrote: Can someone in this group help me with setting up a call center using opensips as the proxy server and asteri

Re: [OpenSIPS-Users] Call Center error

or freeswitch as the media server. On Fri, 19 Apr 2024 at 14:45, Prathibha B wrote: > Can someone in this group help me with setting up a call center using > opensips as the proxy server and asterisk as media server? > > On Fri, 19 Apr 2024 at 10:04, Prathibha B > wrote: &g

Re: [OpenSIPS-Users] Call Center error

Can someone in this group help me with setting up a call center using opensips as the proxy server and asterisk as media server? On Fri, 19 Apr 2024 at 10:04, Prathibha B wrote: > I am getting the error "Cannot handle call" > > On Thu, 11 Apr 2024 at 16:16, Prathibha B >

Re: [OpenSIPS-Users] Call Center error

I am getting the error "Cannot handle call" On Thu, 11 Apr 2024 at 16:16, Prathibha B wrote: > I changed the message_queue uri in cc_flows table to sip:1...@bp.erss.in. > Now getting the following errors in syslog: > > ERROR:nathelper:fix_nated_contact_f: SCRIP

[OpenSIPS-Users] send failed

You tried t make a video call. send failed(477/TM). I am getting this error frequently. -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Call Center error

ERROR:tm:_reply_light: failed to generate 408 reply when a final 407 was sent out ERROR:b2b_entities:_b2b_send_reply: failed to send reply with tm ERROR:b2b_logic:_b2b_handle_reply: Sending reply failed - 408, [B2B.394.162.1712831164.913185781] On Thu, 11 Apr 2024 at 15:33, Prathibha B wrote: > w

Re: [OpenSIPS-Users] Call Center error

"Ref": 0, "Loged in": "NO" } ] } I've logged in 101001 and 101002 in the browser. But the Loged in status is No for both users. On Thu, 11 Apr 2024 at 15:10, Prathibha B wrote: > Cahnged it to $tU, Still getting error. > > On Thu, 11 A

Re: [OpenSIPS-Users] Call Center error

Cahnged it to $tU, Still getting error. On Thu, 11 Apr 2024 at 15:09, Alain Bieuzent wrote: > Hi, > > > > Something wrong about that part for me : ($(tU) == "112") > > Should be : ($tU == "112") > > > > Regards > > > >

Re: [OpenSIPS-Users] Call Center error

_mod: failed to send reply with tm module On Thu, 11 Apr 2024 at 14:30, Prathibha B wrote: > bp.erss.in - asterisk > bp.erss.in:1443 - opensips > > On Thu, 11 Apr 2024 at 14:29, Prathibha B > wrote: > >> *I've created the entries in cc_agents and cc_

Re: [OpenSIPS-Users] Call Center error

bp.erss.in - asterisk bp.erss.in:1443 - opensips On Thu, 11 Apr 2024 at 14:29, Prathibha B wrote: > *I've created the entries in cc_agents and cc_flows table.* > > *cc_age

Re: [OpenSIPS-Users] Call Center error

| ++-+--+-++-+---+--+---+-+-+++-+ On Thu, 11 Apr 2024 at 14:27, Prathibha B wrote: > Getting the following er

[OpenSIPS-Users] Call Center error

Getting the following error in call center module ERROR:call_center:set_call_leg: failed to init new b2bua call (empty ID received) Apr 11 14:06:28 etg-virtual-machine /usr/sbin/opensips[1249949]: ERROR:call_center:w_handle_call: failed to set new destination for call Call center code in

Re: [OpenSIPS-Users] Packet analysis using wireshark

efore running the WebRTC client and it *should* dump the secrets there. > Which will ultimately get picked up by Wireshark and the traffic will > decode. > > Good luck! :) > > Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS Summit 2024 Valencia, May

[OpenSIPS-Users] Packet analysis using wireshark

I am unable to see the Voip calls in wireshark. For signaling opensips is used. The calls are encrypted and it is webrtc communication. Sent from Outlook for Android ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] Continously ringing

User A makes a call to user B. User B didn't accept the call. User B stopped ringing after 1 minute. But User A continues to be in the ringing state until it is manually canceled. Sent from Outlook for Android ___ Users mailing

Re: [OpenSIPS-Users] external applications

will accounting module help here? On Sat, 6 Apr 2024 at 13:04, Prathibha B wrote: > When will the dialog status be in unconfirmed state? I've checked it the > initial state of the call before getting any response. It returns Early > state if the state is trying. But never returns un

Re: [OpenSIPS-Users] external applications

ysg> From: Prathibha B Sent: Saturday, April 6, 2024 10:36:05 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications PFA On Sat, 6 Apr 2024 at 07:14, Prathibha B mailto:prathibhab@gmail.com>> wrote: If someone attempts to make a call and cancel

Re: [OpenSIPS-Users] external applications

PFA On Sat, 6 Apr 2024 at 07:14, Prathibha B wrote: > If someone attempts to make a call and cancels it, I need to maintain a > log of those calls for reporting purposes. > > Sent from Outlook for Android <https://aka.ms/AAb9ysg> > -- >

Re: [OpenSIPS-Users] external applications

a new call. Whatever you are trying to “capture” can be captured there. If you cannot identify the path a new call will take through your script, I don’t think anyone here will be able to help you much, at least not without the entire script. Ben Newlin From: Users on behalf of Prath

Re: [OpenSIPS-Users] external applications

route { if($DLG_status == 1) xlog("UNCONFIRMED\n"); } On Fri, 5 Apr 2024 at 13:09, Prathibha B wrote: > I am not getting $DLG_status = UNCONFORMED at the start of the call. Can > someone help? > > On Fri, 5 Apr 2024 at 05:51, Prathibha B wrote: > >> O

Re: [OpenSIPS-Users] external applications

I am not getting $DLG_status = UNCONFORMED at the start of the call. Can someone help? On Fri, 5 Apr 2024 at 05:51, Prathibha B wrote: > Opensips exec() is working using the absolute path. > > Sent from Outlook for Android <https://aka.ms/AAb9ysg> >

Re: [OpenSIPS-Users] external applications

Opensips exec() is working using the absolute path. Sent from Outlook for Android<https://aka.ms/AAb9ysg> From: Prathibha B Sent: Friday, April 5, 2024 5:49:37 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applications My qu

Re: [OpenSIPS-Users] external applications

g/docs/modules/3.4.x/exec.html#func_exec [2] - https://www.opensips.org/Documentation/Script-CoreParameters-3-4#wdir Ben Newlin From: Users on behalf of Prathibha B Date: Thursday, April 4, 2024 at 11:32 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] external applic

Re: [OpenSIPS-Users] external applications

what does the $rm = SUBSCRIBE mean? On Thu, 4 Apr 2024 at 12:18, Prathibha B wrote: > However this code is getting executed after reaching the ringing state. > > On Thu, 4 Apr 2024 at 12:13, Prathibha B wrote: > >> I am not getting the log also. >> >> On Thu, 4

Re: [OpenSIPS-Users] external applications

However this code is getting executed after reaching the ringing state. On Thu, 4 Apr 2024 at 12:13, Prathibha B wrote: > I am not getting the log also. > > On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > >> if($rm == "INVITE") { >> xlog("Request meth

Re: [OpenSIPS-Users] external applications

I am not getting the log also. On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > if($rm == "INVITE") { > xlog("Request method = $rm"); > exec("script.sh \"INVITE\""); > } > > > With the above code , I am getting the Request method =

Re: [OpenSIPS-Users] external applications

I've used the above code inside the route block. On Thu, 4 Apr 2024 at 12:02, Prathibha B wrote: > if($rm == "INVITE") { > xlog("Request method = $rm"); > exec("script.sh \"INVITE\""); > } > > > With the above code , I am getting the

Re: [OpenSIPS-Users] external applications

if($rm == "INVITE") { xlog("Request method = $rm"); exec("script.sh \"INVITE\""); } With the above code , I am getting the Request method = INVITE twice in the log file. But the exec() is not getting executed. On Wed, 3 Apr 2024 at 19:43, Prathibha B

Re: [OpenSIPS-Users] external applications

celled should result in a state change event – to CANCELLED I think - so I assume you mean you are cancelling it prior to dialog creation, in which case there won’t be any dialog callback. Ben Newlin From: Users on behalf of Prathibha B Date: Wednesday, April 3, 2024 at 2:35 AM To: OpenSIPS

Re: [OpenSIPS-Users] unix timestamp in microseconds

3-4#get_timestamp > > https://www.opensips.org/Documentation/Script-CoreVar-3-4#time > > https://www.opensips.org/Documentation/Script-CoreVar-3-4#Tsm > > > > Please do try to find answers on your own before engaging the mailing list > with such trivial questions. > > &g

[OpenSIPS-Users] location table

Where does opensips store the location details of the end points? I checked in location table in the database. It contains only very few entries and those are very old. -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org

Re: [OpenSIPS-Users] Event ID

I have to pass this event ID to external application. On Wed, 3 Apr 2024 at 16:02, Prathibha B wrote: > Is it possible to retrieve the event ID passed to the client in opensips? > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha ___

[OpenSIPS-Users] Event ID

Is it possible to retrieve the event ID passed to the client in opensips? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] external applications

> > > *From: *Users on behalf of Prathibha B > > *Date: *Tuesday, April 2, 2024 at 8:05 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > >

[OpenSIPS-Users] Trying for more than 60 secs

modparam("tm", "fr_timeout", 45) modparam("tm", "fr_inv_timeout", 60) These are the tm parameters I've set in opensips. When User B is not available, User keeps on Trying without stopping. -- Regards, B.Prathibha ___ Users mailing list

Re: [OpenSIPS-Users] external applications

I tried is_method("INVITE"), but it is getting called only at the start of RINGING. On Tue, 2 Apr 2024 at 15:09, Prathibha B wrote: > I am able to capture the trying status also. But not getting the START of > the call... > > On Tue, 2 Apr 2024 at 14:59, Prathibha B

Re: [OpenSIPS-Users] external applications

I am able to capture the trying status also. But not getting the START of the call... On Tue, 2 Apr 2024 at 14:59, Prathibha B wrote: > How do I identify the START and TRYING state of the call? > > I am able to capture RINGING, ANSWER and TERMINATED states. > > On Tue, 2 Ap

Re: [OpenSIPS-Users] external applications

How do I identify the START and TRYING state of the call? I am able to capture RINGING, ANSWER and TERMINATED states. On Tue, 2 Apr 2024 at 14:51, Prathibha B wrote: > I tried > event_route[E_DLG_STATE_CHANGED] { > > } > > I am getting syntax error. > > On Tue, 2 Apr

Re: [OpenSIPS-Users] external applications

I tried event_route[E_DLG_STATE_CHANGED] { } I am getting syntax error. On Tue, 2 Apr 2024 at 14:45, Prathibha B wrote: > How to use E_DLG_STATE_CHANGED to identify the start of the call? > > > On Wed, 20 Mar 2024 at 19:46, Ben Newlin wrote: > >> You can als

Re: [OpenSIPS-Users] external applications

cu > *Date: *Wednesday, March 20, 2024 at 10:06 AM > *To: *OpenSIPS users mailling list , Prathibha > B > *Subject: *Re: [OpenSIPS-Users] external applications > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > -- > >

[OpenSIPS-Users] unix timestamp in microseconds

How to get the unix timestamp in microseconds? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Send UDP packets

I've used send() function but the data is transmitted as SIP not as UDP. On Mon, 1 Apr 2024 at 12:04, Prathibha B wrote: > which module and function to use to send UDP packets to external > application from opensips? > > I > > -- > Regards, > B.Prathibha >

[OpenSIPS-Users] Send UDP packets

which module and function to use to send UDP packets to external application from opensips? I -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] dropped calls

How to identify the dropped calls in opensips configuration script? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Send failed(477/TM)

You tried t make a video call. send failed(477/TM). I am getting this error frequently. -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] $DLG_dir

$DLG_dir returns null, when used in is_method("INVITE") -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Reg media server

Is it madatorily required to use a media server such as rtpengine. I'm using coturn server as the TURN server for media relay. I'm getting the media through coturn server. Still rtpengine is required? -- Regards, B.Prathibha ___ Users mailing list

Re: [OpenSIPS-Users] variables not working as expected

unable to access the script variable in the route block On Sat, 23 Mar 2024 at 15:39, Prathibha B wrote: > if ( !create_dialog("B") ) { > send_reply(500,"Internal Server Error"); > exit; > } > else > dlg_

[OpenSIPS-Users] variables not working as expected

if ( !create_dialog("B") ) { send_reply(500,"Internal Server Error"); exit; } else dlg_on_answer("dlg_answered"); route[dlg_answered] { $var(payload) = "{'type':'ANSWERED', 'from':" + $fU +", 'to':" + $tU+ "}"; } *Result:*

Re: [OpenSIPS-Users] SDP

ms could be done on the Kamailio SIP router I need to change the a=sendrecv field and it seems to be impossible ☹ Regards … _ Stefan Carlsson From: Users <mailto:users-boun...@lists.opensips.org> On Behalf Of Prathibha B Sent: Thursday

Re: [OpenSIPS-Users] fetch username from the URI

Issue resolved Sent from Outlook for Android<https://aka.ms/AAb9ysg> From: Prathibha B Sent: Thursday, March 21, 2024 5:36:47 PM To: OpenSIPS users mailling list Subject: fetch username from the URI I am getting the source and destination URI from $fu a

[OpenSIPS-Users] fetch username from the URI

I am getting the source and destination URI from $fu and $ru. How to fetch the SIP username from the source and destination URI. What I've tried and not working: # Extract destination URI from the R-URI (Request-URI) $var(destination_uri) = $(ru); # Find the position of

Re: [OpenSIPS-Users] call center

I want to pass the call START, CONNECT, END messages to the external application Sent from Outlook for Android<https://aka.ms/AAb9ysg> From: Prathibha B Sent: Wednesday, March 20, 2024 7:56:14 PM To: Bogdan-Andrei Iancu ; OpenSIPS users mailling list S

Re: [OpenSIPS-Users] call center

users mailling list ; Prathibha B Subject: Re: [OpenSIPS-Users] call center Could you be more specific on your question, like what OpenSIPS should do for the call center application?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siph

Re: [OpenSIPS-Users] external applications

sips.org/html/docs/modules/3.2.x/exec.html > > > On 3/20/2024 5:00 AM, Prathibha B wrote: > > How to integrate OpenSIPS with external applications? > > -- > Regards, > B.Prathibha > > ___ > Users mailing > listUsers@li

[OpenSIPS-Users] external applications

How to integrate OpenSIPS with external applications? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] CDR

How to get the Call Detail Record in Opensips? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Management Interface

What is the alternative to Asterisk Management Interface in OpenSIPS? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] call center

How to integrate Opensips with call center application? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Incoming Call

This happens once in a minutes. Once a minute, the reregistration happens and the page reloads itself. On Sun, 17 Mar 2024 at 18:31, Prathibha B wrote: > The call is generate once in a minute. > > On Sun, 17 Mar 2024 at 18:22, Prathibha B > wrote: > >> Automatically the

Re: [OpenSIPS-Users] Incoming Call

The call is generate once in a minute. On Sun, 17 Mar 2024 at 18:22, Prathibha B wrote: > Automatically the web page with browser phone url initiates the call > without me loading the webpage. > > Attachment: opensips.cfg > > -- > Regards, > B.Prathibha >

[OpenSIPS-Users] Incoming Call

Automatically the web page with browser phone url initiates the call without me loading the webpage. Attachment: opensips.cfg -- Regards, B.Prathibha opensips.cfg Description: Binary data ___ Users mailing list Users@lists.opensips.org

[OpenSIPS-Users] SDP

How to see the SDP in opensips? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] opensips database

How to use encrypted password in opensips? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

I want to setup a call center. Sent from Outlook for Android From: Users on behalf of Alexey Sent: Friday, February 9, 2024 12:24:11 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

[OpenSIPS-Users] OpenSIPS - Asterisk Integration

Has anyone done the integration of Opensips 3.4 with Asterisk 18.20? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Ringing time

How to increase the ringing time? Sent from Outlook for Android ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Dialplan

> https://www.opensips-solutions.com > https://www.siphub.com > > On 05.12.2023 12:49, Prathibha B wrote: > > Yes, you are right. > > On Tue, 5 Dec 2023 at 15:04, Bogdan-Andrei Iancu > wrote: > >> What do you mean by "free call taker in the grou

Re: [OpenSIPS-Users] Dialplan

.com > https://www.siphub.com > > On 04.12.2023 14:30, Prathibha B wrote: > > How to identify a free call taker in the group? > > On Mon, 4 Dec 2023 at 17:57, Prathibha B wrote: > >> How to write dialplan to route the calls coming to 112, to one of the >> free cal

Re: [OpenSIPS-Users] Dialplan

How to identify a free call taker in the group? On Mon, 4 Dec 2023 at 17:57, Prathibha B wrote: > How to write dialplan to route the calls coming to 112, to one of the free > call takers? > > -- > Regards, > B.Prathibha > --

[OpenSIPS-Users] Dialplan

How to write dialplan to route the calls coming to 112, to one of the free call takers? -- Regards, B.Prathibha ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] not getting audio and video

] ERROR:proto_wss:tls_blocking_write: TLS failed to send data On Mon, 2 Oct 2023 at 17:16, Prathibha B wrote: > I'm using opensips-2.4. Connection is getting established between two > browsers. But there is no audio and video. One user is behind NAT. I'm > using rtpengine as media proxy. > >

[OpenSIPS-Users] not getting audio and video

I'm using opensips-2.4. Connection is getting established between two browsers. But there is no audio and video. One user is behind NAT. I'm using rtpengine as media proxy. There is audio and video when both the users are in the same n/w. -- Regards, B.Prathibha

[OpenSIPS-Users] TLS connection errors

On starting up openips, I get TLS connection to 14.139.183.221:63931 write failed (5:-1:9) ERROR:tls_openssl:openssl_tls_write: TLS write error: ERROR:tls_openssl:openssl_tls_blocking_write: TLS failed to send data ERROR:tls_openssl:openssl_tls_write: TLS connection to 111.92.11.139:54102 write

[OpenSIPS-Users] unexpect EOF while reading

opensips version: 3.3 Error: I am using websocket. I am using the letsencrypt certifcate. When I connected from web to web and when verify_cert is 1 and require_cert is 1, get the following errors: ERROR:tls_openssl:openssl_tls_accept: New TLS connection from 14.139.183.221:53946 failed to

Re: [OpenSIPS-Users] Temporarily unavailable

When user2 accepts the call, the above error occurs. On Tue, 26 Sept 2023 at 14:08, Prathibha B wrote: > I'm getting this error in log: > > New TLS connection from 45.33.68.226:42316 failed to accept > ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0AEA:SSL > rou

Re: [OpenSIPS-Users] Temporarily unavailable

I'm getting this error in log: New TLS connection from 45.33.68.226:42316 failed to accept ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0AEA:SSL routines::callback failed ERROR:proto_wss:wss_read_req: cannot fix read connection On Tue, 26 Sept 2023 at 13:41, Prathibha B wrote

[OpenSIPS-Users] Temporarily unavailable

I've created two users and they are registered to opensips. When I try to connect, user 1 to user 2, I get temporarily unavailable. Config file is attached with this email. -- Regards, B.Prathibha # # OpenSIPS residential configuration script # by OpenSIPS Solutions # # Please refer to the

Re: [OpenSIPS-Users] Temporarily unavailable

I am using websocket. I am trying web to web call. On Tue, 26 Sept 2023 at 13:40, Prathibha B wrote: > I've created two users and they are registered to opensips. When I try to > connect, user 1 to user 2, I get temporarily unavailable. Config file is > attached with t

[OpenSIPS-Users] SSL Certificate error

TLSv1.2 (OUT), TLS header, Supplemental data (23): * OpenSSL SSL_write: Connection reset by peer, errno 104 * Failed sending HTTP request * Connection #0 to host bp3.erss.in left intact curl: (55) OpenSSL SSL_write: Connection reset by peer, errno 104 -- Regards, B.Prathibha

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