Re: [SR-Users] WS to WS calls - No ACK received for 200OK

ack: failure to add branches Jun 21 19:07:15 acstemplate /usr/sbin/kamailio[7869]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Thanks & Regards, Amit Patkar On 6/21/2016 5:50 PM, Nagorny, Dimitry wrote: Hi Amit, I had the

[SR-Users] WS to WS calls - No ACK received for 200OK

Hi I am using Kamailio as Websocket proxy. User 1 & User 2 are registered on Kamailio over WebSocket. When User 1 calls User 2, User 2 gets ring and answers the call. 200 OK message is received by User 1 but ACK response sent by User 1 does not reach User 2. Since User 2 didn't get ACS, after

[SR-Users] Dispatcher - Route call to alternate server if there is delay to answered call

work congestion), I wish to send this call to M2. What are the options to achieve it? Thanks & Regards, Amit Patkar ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org

Re: [SR-Users] Ways to reload Kamailio configuration file without restart

Thanks Daniel. Since there are WebRTC clients connected, restart should be avoided in my case Is there any plan to add configuration file reload in future? *Thanks & Regards,* Amit Patkar On 1/8/2016 2:07 PM, Daniel-Constantin Mierla wrote: Hello, if you need to change the routing l

Re: [SR-Users] rtpengine - Error when sending message. Error: Invalid argument

Thanks Richard. It worked. *Thanks & Regards,* Amit On 2/14/2015 7:02 PM, Richard Fuchs wrote: On 14/02/15 08:13 AM, Amit Patkar wrote: Hi I am getting error with rtpengine. Running Kamailio 4.2.3 I am trying to call from conventional SIP client to WebRTC client Google Ch

[SR-Users] rtpengine - Error when sending message. Error: Invalid argument

Hi I am getting error with rtpengine. Running Kamailio 4.2.3 I am trying to call from conventional SIP client to WebRTC client Google Chrome v38.0.2125.104 Firefox v33.0 Using sipML5 as WebRTC client root@rtcpbx:/home/avhan# /usr/sbin/rtpengine --interface=127.0.0.1\!192.168.2.161 --listen-n

Re: [SR-Users] No audio when connect from public network but it works for lan users

What is the value set to localnet and externip parameters? NAT will not work without setting these parameters. These are global parameters. Regards, Amit Patkar___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.si

Re: [SR-Users] No audio when connect from public network but it works for lan users

Please check sip.conf. You need to enable NAT options. Asterisk need to publish public IP in SDP for RTP traffic to reach your network. Asterisk need to differentiate local clients and external clients. localnet and externip parameters should be configured correctly. Regards, Amit Patkar On

Re: [SR-Users] Issue with 4.2.0 and nathelper and/or rtpproxy

Hi You should compare this line in your configuration file. if (nat_uac_test("*18*")) { It may be using different parameters. *Regards,* Amit On 11/18/2014 5:23 PM, Igor Potjevlesch wrote: Hello, I can reproduce the issue on a pre-production system. So, I downgraded to 4.1.5. Here is t

Re: [SR-Users] One sided or no voice issue with websockets

Please check ICE server settings. Your browser may be publishing local IP. One way voice or no voice is typical case when client is behind firewall. Regards, Amit Patkar dodul wrote: >Hi > >I didn't get any responses from anyone regarding my issue. Can someone please >give

[SR-Users] Mozilla Firefox + SIP Phone voice not established

ee4:RTCPd7:packetsi2e5:bytesi56e6:errorsi0eee6:result2:oke Oct 16 18:39:20 kamailio /usr/local/sbin/kamailio[28815]: NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1413464960;method=BYE;from_tag=2c76015a;to_tag=R3okhRsDImBAiNMCZ04e;call_id=1b211056-eb41-7edf-59