Re: [OpenSIPS-Users] regular expression matching question

ause of a typo in the syntax. You have used the ~= operator instead > of the =~ operator, which is the correct way to test if a string matches > a regular expression in PCRE. > > Op di 2 jan 2024 om 12:28 schreef Faheem Muhammad : > >> Hi everyone, >> >> Firstly, I wish

[OpenSIPS-Users] regular expression matching question

Hi everyone, Firstly, I wish you all a Happy New Year with the best wishes for each community member and the OpenSIPS project. I have a question related to regular expressions matching in an IF condition. There are two identical regular expressions. One RE is matching and one RE is not matching t

Re: [OpenSIPS-Users] Monitoring end to end solution.

, Faheem Muhammad wrote: > No, with SIPp you can create any type of call scenario including REGISTER > + INVITE scenarios. > Follow the below mentioned blog entry 'Scenario 2'. > I posted it few years back. Unfortunately original domain is no more > available, but arc

Re: [OpenSIPS-Users] Monitoring end to end solution.

No, with SIPp you can create any type of call scenario including REGISTER + INVITE scenarios. Follow the below mentioned blog entry 'Scenario 2'. I posted it few years back. Unfortunately original domain is no more available, but archive link works. http://web.archive.org/web/20130924010913/http:/

Re: [OpenSIPS-Users] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()

Husnain, The same type of question is already answered by Kristian F. Høgh on the opensips mailing list. Try to search "*[OpenSIPS-Users] Playing caller a file before dialing callee*" Hope it will solve your problem. Faheem On Thu, Jan 7, 2016 at 10:37 AM, Hamid Hashmi wrote: > Try the follow

Re: [OpenSIPS-Users] SIP Messenger for linux

You can try Blink phone. It supports SIP SIMPLE. Faheem On Tue, Jan 5, 2016 at 5:04 PM, Hamid Hashmi wrote: > Is there any opensource sip messenger for linux to test IM on SIP ? > > *Hamid R. Hashmi* > Software Engineer - VoIP > Vopium A/S > > ___ > U

[OpenSIPS-Users] wrong contact header in reply route

Hi, I'm facing an ACK loop while making calls. The issue is occurred due to bad CONTACT header. Scenario is as follows. sip_user1(10.111.11.136) <-> OpenSIPS-1(10.109.5.189) <-->Outbound_proxy1 (10.109.5.237) The issue occurs when sip_user1 sends INVITE to OpenSIPS-1. OpenSIPS-1 processes the req

Re: [OpenSIPS-Users] What information about calls are kept by OpenSIPS and how to handle it?

You can use acc module for #1 and 2 http://www.opensips.org/html/docs/modules/1.6.x/acc It does all type of call logs. you can find the details in the documentation. Regarding #3, you can filter out different soft phone using predefined $ua (useragent field), but this UAC defined field can be mani