Re: [OpenSIPS-Users] sip-i ?

Hi Vlad So I actually want User service info to look like your code: User Service Information Mandatory Parameter: User service information (29) Pointer to Parameter: 3 Parameter Length: 3 User service information (-> Q.931 Bearer_capability): 8090a2

Re: [OpenSIPS-Users] sip-i ?

I’m working on adding proper ANSI ISUP support to opensips. Sent from my iPhone > On Mar 26, 2019, at 10:18 AM, Vlad Patrascu wrote: > > Hi Sean, > > You can set a custom value for any ISUP parameter that appears in the > specification. In your case, you can do: > > $isup_param(User

Re: [OpenSIPS-Users] sip-i ?

Hi Sean, You can set a custom value for any ISUP parameter that appears in the specification. In your case, you can do: $isup_param(User Service Information) = "0x8090A2" Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 03/26/2019 04:22 PM, Sean Watkins wrote:

Re: [OpenSIPS-Users] sip-i ?

Hi Bogan I got some more success yesterday with it -- I'll post my configs after, for a SIP-I to SIP gateway. Where I'm stuck now - there seems to be some differences between ANSI + ITU. I've got the invite coming into Opensips, then I'm successfully adding the SDP info for ISUP. (Whomever came

Re: [OpenSIPS-Users] sip-i ?

Hi Sean, Where have you got stuck ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 2019 https://www.opensips.org/events/Summit-2019Amsterdam/ On 03/22/2019 10:47 PM, Sean Watkins wrote: Hi Are there any SIP-I experts on

[OpenSIPS-Users] sip-i ?

Hi Are there any SIP-I experts on here that are willing todo some consulting? I need to get a SIP-I to SIP gateway working... I've got partial call connectivity working, but I don't seem to know what I'm doing.. Don't get all the ins and outs of reply etc.. Sean

Re: [OpenSIPS-Users] SIP-I and CANCEL

On Fri, Dec 7, 2018, 18:45 Vlad Patrascu Hi Giovanni, > > Although I'm not claiming to be an expert at all in isup-sip interworking, > the ITU Q.1912.5 document seems to say that it is not necessary to > encapsulate REL in CANCEL. You can take a look at the example in Figure > III.13/Q.1912.5

Re: [OpenSIPS-Users] SIP-I and CANCEL

Hi Giovanni, Although I'm not claiming to be an expert at all in isup-sip interworking, the ITU Q.1912.5 document seems to say that it is not necessary to encapsulate REL in CANCEL. You can take a look at the example in Figure III.13/Q.1912.5 from Appendix III section of the document.

Re: [OpenSIPS-Users] SIP-I and CANCEL

On Fri, Dec 7, 2018 at 6:14 PM Bogdan-Andrei Iancu wrote: > Hi Giovanni, > > The CANCEL is hop-by-hop, which means each SIP hop will generate a > completely new CANCEL (to be sent further) - there is not proxying, so the > changes you do over the incoming CANCEL will not propagate into the >

Re: [OpenSIPS-Users] SIP-I and CANCEL

Hi Giovanni, The CANCEL is hop-by-hop, which means each SIP hop will generate a completely new CANCEL (to be sent further) - there is not proxying, so the changes you do over the incoming CANCEL will not propagate into the outgoing CANCEL. I cannot recall it, but is there a standard case of

[OpenSIPS-Users] SIP-I and CANCEL

Hi all, using 2.4 I am trying to add_isup_part("Release"); to a CANCEL It does not give errors (I use same isup_param(s) as per BYE), but the CANCEL that is being sent out does not have the isup part. Eg, it does not even have a body, and Content-Length is 0. I suspect t_relay, when it

Re: [OpenSIPS-Users] SIP-I Calling Party number being changed

Thanks for the explanation. We changed + to 000 now things are pretty good. Thanks. Virus-free. www.avast.com

Re: [OpenSIPS-Users] SIP-I Calling Party number being changed

Hi, I'm not an expert on ITU-T standards but from what I know, the "+" sign is only an indication that an international prefix is required and thus cannot be set as a digit in the ISUP number related parameters. Being an unsupported character, the sip_i module wrote it as 0. Maybe a warning

[OpenSIPS-Users] SIP-I Calling Party number being changed

Greetings list, I am using the sip-i module for conversion between sip to sip-i. I want to set calling party number and called party number in e164 format. Below is how my code looks like. route{ ... xlog("Here is Called party number: $rU"); xlog("Here is Calling party number: