OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server.
OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server.
Questions tagged [opensips]
133 questions
0
votes
1 answer
how subscribe to buddylist using resource list xml
I need to implement a SIP subscription to a resource list, under which tag I should put the XML buddy list that contains the desired SIP ID (to monitor their states). Note that I'm using Jain-SIP API, and I implemented the single subscription and is…
Salim R
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0
votes
1 answer
how to have Asterisk-AGI like functionality in OpenSIPs or Kamailio
I used to send AGI requests from Asterisk SIP server to an external App and reply back with commands like DIAL(...). I'm trying to do the same in opensips using Events Interface (UDP), Management Interface (UDP too) and the dialog module. Any advice…
Mohammad Qandeel
- 585
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- 11
0
votes
1 answer
drop a packet before the start of routing call: openSIPS
I want to drop the sip packets which have to and contact field empty.
INVITE sip:******************* SIP/2.0
Record-Route:
Via: SIP/2.0/UDP *********;branch=z9hG4bK0e44.f7bd2db2.0
Via: SIP/2.0/UDP…
pratik
- 46
- 9
0
votes
1 answer
Open source solution for SIP based video streaming server?
I need to implement a SIP based video streaming server, which can communicate with SIP cameras as well as SIP clients on iOS/Android.
I'm investigating open source projects openSIPS and Kamailio, just want to know which one is more suitable for this…
ciphor
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0
votes
1 answer
Information for connecting SIP user Agent with SIP server(Opensip)
I am a newbie to sip server and i am experimenting with it using c programming.
I have installed open sip server on my ubuntu 13.04.
and i am trying to configure it.
can somebody provide me a basic example showing how a user agent registers to sip…
GP007
- 691
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0
votes
1 answer
OpenSIP not sending cancel to UAS in case it received 200 from UAC. Verified in 1.7.2 and 1.8
SIP Call Graph Diagram when Bug comes:
A = UAC
B = OpenSIPS
C = UAS
A ---------- INVITE ---------> B
A <-------- STATUS 100 TRYING ------- B
B ---------- INVITE ---------> C
B <-------- STATUS 100 TRYING --------- C
B <-------- STATUS 200 OK…
Mani
- 508
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0
votes
2 answers
opensips open ims and asterisk configuration for audio video calls on Ubuntu?
I am not sure if it is a correct place for such question but unfortunately I did not find any other stackexchange site to ask this question. But I have read some similar question here like on Open IMS and on Asterisk.
My Question is, I want to make…
Abdul Rehman
- 2,224
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-1
votes
1 answer
1 way audio only when registering to OpenSIPs in front of Asterisk
Long time Asterisk user but fairly new to OpenSIPs. I have a SIP phone working with audio both directions when registering to and receiving calls directly from Asterisk. The same phone works with 2 way audio if I register to OpenSIPs and receive a…
Fonewiz
- 2,065
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- 17
-1
votes
1 answer
Asterisk 13 PJSIP sometime sound coming sometime not coming
I recently set up my asterisk 13 with PJSIP and database. All working fine, but sometimes I get no voice, where most of the time I get a voice. So I need RTP software? following is detail log, I am looking but not found any voice or codec issue, as…
Kamal Panhwar
- 2,345
- 3
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- 37
-1
votes
1 answer
regex match equal with tilde Vs double equal sign
Its wired question but i want to know what is the difference between =~ and ==
Following "string" i am trying to find.
if($ua =~ "friendly-scanner") {
drop()
}
Vs
if($ua == "friendly-scanner") {
drop()
}
Satish
- 16,544
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- 149
-2
votes
1 answer
asterisk to opensips conversion. all help appresciated
m curently working on converting an esxisting asterisk server to opensips, for better perfomance
for the most part it is working, but ive encountered an issue i cant really figure out.
asterisk is doing this :
if ("${fromourmobile}" != "") // Check…
Claudi
- 125
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- 9
-3
votes
1 answer
Getting "No such commands sip reload"
I configured sip.conf and extensions.conf files, I am Able to reload the dialplan but whenever I type sip reload, sip show users or sip show peers, I am getting no such commands, type 'core sip show help sip reload',...
Karthick29
- 1
- 1
- 1
-4
votes
1 answer
How to connect X-Lite softphone from host to guest vm with asterisk?
I am desperate. I've install asterisk on vm 1 (centos) and opensips on vm2(centos), and everything works well so far. Now I need to connect softphone from host to vm1 (to make a call (I'm traying to set up auto-dial out system))) and don't know how…
