OverSIP with PortaOne is not working

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Yasir Ikram

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Jan 31, 2018, 5:30:34 AM1/31/18
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Hi,

I am working on a Browser based SIP client. I have a SIP server PortaOne which doesn't support Web Sockets. So I have decided to use SIP-Proxy.

I have installed OverSIP on my linux ubuntu 14.04 server.  and for WebRTC SIP client I am using JSSIP.

I register SIP accounts successfully but when I call from my web browser it successfully dial the call but it disconnect the call when call picked up on destination.

Please guide me if there is any configuration required in oversip to resolve this issue.


Thanks

José Luis Millán

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Jan 31, 2018, 6:48:15 AM1/31/18
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Hi Yasir,

You are not giving any information at all. If the call disconnects, have you checked the logs in the SIP client or in the SIP server, or even in OverSIP?

Help yourself by checking the logs in the endpoints first (SIP client & Server). I doubt there is an issue with the proxy (OverSIP) here but it's impossible to know since no data is provided.

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José Luis Millán

Iñaki Baz Castillo

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Jan 31, 2018, 6:53:45 AM1/31/18
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Does PortaOne support WebRTC media? I don't mean WebSocket but WebRTC media (ICE, DTLS, SRTP).
Please, confirm. 

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Yasir Ikram

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Feb 1, 2018, 11:07:29 AM2/1/18
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Yes PortaONE support (ICE, DTLS, SRTP)


On Wednesday, January 31, 2018 at 4:53:45 PM UTC+5, Iñaki Baz Castillo wrote:
Does PortaOne support WebRTC media? I don't mean WebSocket but WebRTC media (ICE, DTLS, SRTP).
Please, confirm. 
El 31 ene. 2018 11:30, "Yasir Ikram" <mail....@gmail.com> escribió:
Hi,

I am working on a Browser based SIP client. I have a SIP server PortaOne which doesn't support Web Sockets. So I have decided to use SIP-Proxy.

I have installed OverSIP on my linux ubuntu 14.04 server.  and for WebRTC SIP client I am using JSSIP.

I register SIP accounts successfully but when I call from my web browser it successfully dial the call but it disconnect the call when call picked up on destination.

Please guide me if there is any configuration required in oversip to resolve this issue.


Thanks

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Yasir Ikram

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Feb 1, 2018, 11:17:23 AM2/1/18
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Hi,

Thank you for your response. Please see below OverSIP logs of a particular call which is disconnected when picked up on destination:

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <SIP Request 9001360> applying outgoing Outbound support

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 9001360

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <SIP Request 9001360> replying 100 "Trying"

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <WsFraming> sending text frame: payload_length=267

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <Proxy proxy_out 9001360> destination found in the DNS cache

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <Proxy proxy_out 9001360> trying single target: udp:78.40.244.5:5060

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <Proxy proxy_out 9001360> received response 180

Jan 30 06:51:59 s112209 oversip[26917]:   INFO: <SipEvents> [user] on_provisional_response: 180 'Ringing'

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <SIP Request 9001360> forwarding response 180 "Ringing"

Jan 30 06:51:59 s112209 oversip[26917]:  DEBUG: <WsFraming> sending text frame: payload_length=616

Jan 30 06:52:03 s112209 oversip[26917]:  DEBUG: <NICT 8cc66e4f9f2752160186063b0a316ba3da09c31e> timer K expires, transaction terminated

Jan 30 06:52:04 s112209 oversip[26917]:  DEBUG: <NICT 754089bd9f2752160186063b0a316ba3da09c31e> timer K expires, transaction terminated

Jan 30 06:52:14 s112209 oversip[26917]:  DEBUG: <ICT 69e8fce34b152552f7bd81de1af30d89da6ca49e> sending ACK for [3456]XX response

Jan 30 06:52:14 s112209 oversip[26917]:  DEBUG: <Proxy proxy_out 9001360> received response 500

Jan 30 06:52:14 s112209 oversip[26917]:   INFO: <SipEvents> [user] on_failure_response: 500 'Expired'

Jan 30 06:52:14 s112209 oversip[26917]:  DEBUG: <SIP Request 9001360> forwarding response 500 "Expired"

Jan 30 06:52:14 s112209 oversip[26917]:  DEBUG: <WsFraming> sending text frame: payload_length=585

Jan 30 06:52:15 s112209 oversip[26917]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=411

Jan 30 06:52:15 s112209 oversip[26917]:  DEBUG: <WsSipApp> received WS message: type=text, length=411

Jan 30 06:52:15 s112209 oversip[26917]:  DEBUG: <IST 9001360> ACK received during completed state, now confirmed

Jan 30 06:52:16 s112209 oversip[26917]:   INFO: <SipEvents> [user] INVITE from sip:20...@107.150.61.226 (UA: sipcli/v1.8) to sip:4309441923




On Wednesday, January 31, 2018 at 4:48:15 PM UTC+5, José Luis Millán wrote:
Hi Yasir,

You are not giving any information at all. If the call disconnects, have you checked the logs in the SIP client or in the SIP server, or even in OverSIP?

Help yourself by checking the logs in the endpoints first (SIP client & Server). I doubt there is an issue with the proxy (OverSIP) here but it's impossible to know since no data is provided.
2018-01-31 11:30 GMT+01:00 Yasir Ikram <mail....@gmail.com>:
Hi,

I am working on a Browser based SIP client. I have a SIP server PortaOne which doesn't support Web Sockets. So I have decided to use SIP-Proxy.

I have installed OverSIP on my linux ubuntu 14.04 server.  and for WebRTC SIP client I am using JSSIP.

I register SIP accounts successfully but when I call from my web browser it successfully dial the call but it disconnect the call when call picked up on destination.

Please guide me if there is any configuration required in oversip to resolve this issue.


Thanks

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José Luis Millán

Iñaki Baz Castillo

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Feb 1, 2018, 11:19:34 AM2/1/18
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Rather than OverSIP logs, check your SIP PBX/server logs since it's
the one reply SIP 500 error:

> forwarding response 500 "Expired"
Iñaki Baz Castillo
<i...@aliax.net>
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