Re: [asterisk-dev] Mailing List Future
Hello, Am Dienstag, dem 12.12.2023 um 12:30 -0400 schrieb Joshua C. Colp: > On Tue, Dec 12, 2023 at 12:10 PM Henning Westerholt > wrote: > > Hello, > > > > > > > > the majority of the responses seems to be against the > > discontinuation of the mailing list. > > > > > > Yes, this is true. It would be nice to have more input though so if > other individuals have opinions (including why they want the lists to > continue and what they use them for) then that would be beneficial. > I'm talking to the over 2100 people who haven't responded to this > thread. I also prefer to read a mailing list. Querying forums or github discussions is not a solution for me. Just my 2 cents as a long time follower of this list. (One of the 2100). > > > Has a decision already been made in the last week? Or is this still > > discussed internally or with some of the people that offered help > > in keeping the mailing lists running? > > > > > > The only decision that has been made is that the instance of > lists.digium.com as it exists today will go away. I don't expect any > explicit decision to occur until next year. I'm personally going on > vacation for 2 weeks, though others may continue any discussion in my > absence to get further insight. Have a nice vacation! Karsten Wemheuer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip
Hi *, Am Donnerstag, dem 02.03.2023 um 13:05 -0400 schrieb Joshua C. Colp: > On Thu, Mar 2, 2023 at 1:00 PM Karsten Wemheuer wrote: > > Hi Joshua, > > > > Am Donnerstag, dem 02.03.2023 um 12:44 -0400 schrieb Joshua C. > > Colp: > > > On Thu, Mar 2, 2023 at 12:37 PM Karsten Wemheuer > > > wrote: > > > > Hi Joshua, > > > > > > > > thank You, for answering. > > > > > > > > Am Donnerstag, dem 02.03.2023 um 09:17 -0400 schrieb Joshua C. > > > > Colp: > > > > > On Thu, Mar 2, 2023 at 9:04 AM Karsten Wemheuer > > > > > wrote: > > > > > > Hi *, > > > > > > > > > > > > Maybe I found a small bug or I am doing something wrong. > > > > > > > > > > > > When I do a "Transfer" on a call that arrives via PJSIP, > > > > > > Asterisk sends > > > > > > a "302 Moved Temporarily" to perform the transfer. > > > > > > > > > > What version of Asterisk? What is the precise transport > > > > > configuration? > > > > > > > > As written below it was Version 18. The exact version is > > > > 18.16.0. > > > > > > > > > > In the future please always provide the precise Asterisk version. > > > It's important, as code changes. > > > > > > What is the precise transport configuration in PJSIP? > > > > > > > Transport section is below: > > > > [transport-tcp] > > type = transport > > protocol = tcp > > bind = 0.0.0.0:25060 > > external_media_address = 91.2.166.143 > > external_signaling_address = 91.2.166.143 > > local_net = 10.0.1.0/24 > > local_net = 192.168.10.0/24 > > local_net = 169.254.0.0/24 > > tos = 96 > > allow_reload = no > > > > Then the Contact replacement for NAT purposes may not be specific > enough. File an issue[1] however ALSO include a full SIP trace. > > [1] https://issues.asterisk.org/jira I filed an issue about this. No one has worked on the issue yet, so I would start with this. Can anyone help me get started? [1] https://issues.asterisk.org/jira/browse/ASTERISK-30451 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip
Hi Joshua, Am Donnerstag, dem 02.03.2023 um 12:44 -0400 schrieb Joshua C. Colp: > On Thu, Mar 2, 2023 at 12:37 PM Karsten Wemheuer > wrote: > > Hi Joshua, > > > > thank You, for answering. > > > > Am Donnerstag, dem 02.03.2023 um 09:17 -0400 schrieb Joshua C. > > Colp: > > > On Thu, Mar 2, 2023 at 9:04 AM Karsten Wemheuer > > > wrote: > > > > Hi *, > > > > > > > > > > > > > > > > Maybe I found a small bug or I am doing something wrong. > > > > > > > > > > > > > > > > When I do a "Transfer" on a call that arrives via PJSIP, > > > > Asterisk sends > > > > > > > > a "302 Moved Temporarily" to perform the transfer. > > > > > > What version of Asterisk? What is the precise transport > > > configuration? > > As written below it was Version 18. The exact version is 18.16.0. > > In the future please always provide the precise Asterisk version. > It's important, as code changes. > > What is the precise transport configuration in PJSIP? > Transport section is below: [transport-tcp]type = transportprotocol = tcpbind = 0.0.0.0:25060external_media_address = 91.2.166.143external_signaling_address = 91.2.166.143local_net = 10.0.1.0/24local_net = 192.168.10.0/24local_net = 169.254.0.0/24tos = 96allow_reload = no Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip
Hi Tom, thanks for Your answer. Am Donnerstag, dem 02.03.2023 um 08:18 -0500 schrieb Tom Ray: > On Mar 2, 2023 at 8:05 AM -0500, Karsten Wemheuer , > wrote: > > > Hi *, > > > > > > > > Maybe I found a small bug or I am doing something wrong. > > > > > > > > When I do a "Transfer" on a call that arrives via PJSIP, Asterisk > > sends > > > > a "302 Moved Temporarily" to perform the transfer. > > > > > > For the record, this isn't a transfer it is a redirect. They are > completely different things. The first thing we would need to know is > how you are doing this. Are you immediately using the redirect > features in Asterisk to send back a 302 or is more happening that > results in a 302 being done? Sorry, I mean: I use the dialplan application "transfer" to do a 302 Redirect. With chan_sip it wasTransfer +49xxxThis does not work with pjsip, so I useTransfer sip:+49xxx@ip-addressorTransfer sip: +49...@domain.tld > > Unlike chan_sip, the contact header is set different and maybe > > > > incorrectly with PJSIP: > > > > > > > > chan_sip: > > > > Contact: Transfer > > > > > > > > pjsip: > > > > Contact: > > > > > > We will probably need to see actual SIP debugs and SIP messages so we > can see how this is being sent to the carrier. Doing this in dialplan Transfer I got this with ngrep:T 192.168.10.70:59371 -> 217.0.149.48:5060 [AP] #9SIP/2.0 302 Moved Temporarily.Via: SIP/2.0/TCP 217.0.149.48:5060;rport=5060;received=217.0.149.48;branch=z9hG4bKmavodi -0-264-c43-4-100-4260-5f3047d72-a81--d5- cf87-5f3cccbfe97f2-459902267-5827.Record-Route: .Call-ID: bw171319464020323-1506406...@62.156.74.66.from: ;tag=1594730888-163599464- .To: ;tag=ff7ed316- 6fcf-40e6-ae35-c4077a100999;cscf.CSeq: 294435189 INVITE.Server: Asterisk.Contact: .Reason: Q.850;cause=0.Supported: histinfo.Content-Length: 0. I am using Asterisk 18.16 Is it possible to put a uri with a domain instead of ip address in contact header? The provider gave me an example where the only obvious difference is the portion after the @ sign in the Contact header. Thanks,Karsten > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip
Hi Joshua, thank You, for answering. Am Donnerstag, dem 02.03.2023 um 09:17 -0400 schrieb Joshua C. Colp: > On Thu, Mar 2, 2023 at 9:04 AM Karsten Wemheuer wrote: > > Hi *, > > > > > > > > Maybe I found a small bug or I am doing something wrong. > > > > > > > > When I do a "Transfer" on a call that arrives via PJSIP, Asterisk > > sends > > > > a "302 Moved Temporarily" to perform the transfer. > > What version of Asterisk? What is the precise transport > configuration? As written below it was Version 18. The exact version is 18.16.0. > > > > Unlike chan_sip, the contact header is set different and maybe > > > > incorrectly with PJSIP: > > > > > > > > chan_sip: > > > >Contact: Transfer > > > > > > > > pjsip: > > > >Contact: > > > > > > > > The difference are domain (chan_sip) vs. local IP address (pjsip) > > and > > > > the additional (wrong?) port number. The IP address is the one of > > my > > > > router, but the port number should be 25060, because asterisk is > > > > configured to use this port. > > > > > > > > The transfer works with asterisk 11 and chan_sip. It does not work > > with > > > > pjsip and asterisk 18. My provider does not accept the transfer > > done > > > > with pjsip. Either the provider expects the domain in the contact > > > > header or the error is in the wrong port number. > > > > > > > > Is this a bugf or how to use transfer application in combination > > with > > > > pjsip? > > For questions like this in the future please use either the asterisk- > users mailing list or the community forum[1]. I had used the forum because the user mailing list was not working. Unfortunately the replies were not helpful and I am not sure if it is a bug. How can I make the Contact header contain a URI with domain part (instead of ip address)? Thanks, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip
Hi *, Maybe I found a small bug or I am doing something wrong. When I do a "Transfer" on a call that arrives via PJSIP, Asterisk sends a "302 Moved Temporarily" to perform the transfer. Unlike chan_sip, the contact header is set different and maybe incorrectly with PJSIP: chan_sip: Contact: Transfer pjsip: Contact: The difference are domain (chan_sip) vs. local IP address (pjsip) and the additional (wrong?) port number. The IP address is the one of my router, but the port number should be 25060, because asterisk is configured to use this port. The transfer works with asterisk 11 and chan_sip. It does not work with pjsip and asterisk 18. My provider does not accept the transfer done with pjsip. Either the provider expects the domain in the contact header or the error is in the wrong port number. Is this a bugf or how to use transfer application in combination with pjsip? Thanks Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Using SIP TLS with Mediasec
Hello Michael, thanks a lot! Have a nice day. Karsten Am Montag, dem 30.01.2023 um 19:46 +0100 schrieb Michael Maier: > Hello Karsten, > > attached you will find the patch for 18.16. The patch consists of 4 > files, which should be applied behind one another, starting with 1 > (patch -p1 ...). The first of the attached patches removes the new > Mediasec patch. > > > Thanks > Michael > > > On 30.01.23 at 17:58 Karsten Wemheuer wrote: > > Hi *, > > > > I am currently testing Asterisk 18.16 with TLS and Mediasec. I am > > testing it with Telekom CompanyFlex. The trunk registers and after > > a > > while the request to re-register gets a "403 Forbidden". > > I had previously used Asterisk 18.15 with the patch from Michael > > Maier. > > That worked flawlessly. > > Why are there two approaches to solving the "Mediasec" problem? > > What is > > the obstacle that prevents to use the working patch from Michael > > Maier? > > > > Thanks for clarification. > > > > Have a nice day. > > > > Karsten > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Using SIP TLS with Mediasec
Hi *, I am currently testing Asterisk 18.16 with TLS and Mediasec. I am testing it with Telekom CompanyFlex. The trunk registers and after a while the request to re-register gets a "403 Forbidden". I had previously used Asterisk 18.15 with the patch from Michael Maier. That worked flawlessly. Why are there two approaches to solving the "Mediasec" problem? What is the obstacle that prevents to use the working patch from Michael Maier? Thanks for clarification. Have a nice day. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] pjsip seems to send ACK responses to the wrong destination
Hi Joshua, Am Montag, dem 21.03.2022 um 07:34 -0300 schrieb Joshua C. Colp: > On Mon, Mar 21, 2022 at 7:18 AM Karsten Wemheuer > wrote: > > Hi *, > > > > i am trying to analyze a problem with pjsip. > > > > Scenario: Phones are registered to opensips. From there the calls > > go to > > asterisk and then on via the trunk. This works fine. > > > > In the opposite direction there is sometimes a problem: > > A call comes in over the trunk, asterisk sends the INVITE to > > opensips. > > From there the INVITE goes to the phone. After the call is answered > > (200 OK from phone via proxy), asterisk sends the ACK not via the > > proxy > > but directly to the phone. Looking at the debug log it looks like > > the > > destination address of the ACK is obtained from the Contact or RTP > > data > > and not from the Via header. > > > > I would like to check the source code to see if I am doing > > something > > wrong or if there is a bug. Where do I enter to investigate the > > construction of the ACK packet? > > I would not suggest looking at the source code for this. You would > still have to understand SIP itself to know what is going on, so the > RFC is really the best place. > > For your specific issue - unless the proxy is doing record routing, > then the behavior is correct. The Contact header would be used for > sending the ACK. RTP is never used for SIP signaling destination. > Thanks a lot! The hint with record routing was very helpful. I have it working now. I just was confused by the debug log which lead me in the wrong direction. Have a nice day. Best regards Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] pjsip seems to send ACK responses to the wrong destination
Hi *, i am trying to analyze a problem with pjsip. Scenario: Phones are registered to opensips. From there the calls go to asterisk and then on via the trunk. This works fine. In the opposite direction there is sometimes a problem: A call comes in over the trunk, asterisk sends the INVITE to opensips. From there the INVITE goes to the phone. After the call is answered (200 OK from phone via proxy), asterisk sends the ACK not via the proxy but directly to the phone. Looking at the debug log it looks like the destination address of the ACK is obtained from the Contact or RTP data and not from the Via header. I would like to check the source code to see if I am doing something wrong or if there is a bug. Where do I enter to investigate the construction of the ACK packet? Thanks for any hints, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Encrypted telephony, SIP-TLS, SRTP and mediasec
Hi, Am Mittwoch, dem 16.03.2022 um 08:19 -0300 schrieb Joshua C. Colp: > On Wed, Mar 16, 2022 at 8:16 AM Karsten Wemheuer > wrote: > > Hi, > > > > Deutsche Telekom requires Mediasec attributes for encrypted > > telephony. > > For the Trunk product (business market segment), encryption is > > mandatory if access is provided via another carrier. Since Deutsche > > Telekom is one of the largest VoIP providers in Germany, it would > > be > > nice if Asterisk could be used here. > > > > I know the patch from Michael Maier that was mentioned here on the > > mailing list. I was able to install it successfully together with > > Asterisk 18.10. > > > > Are there any plans to add Mediasec support to Asterisk or to > > include > > this patch? > > I know of noone working on such a patch for inclusion as of this > time. The stated patch can not be included due to licensing. > What is the licensing issue? Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Encrypted telephony, SIP-TLS, SRTP and mediasec
Hi, Deutsche Telekom requires Mediasec attributes for encrypted telephony. For the Trunk product (business market segment), encryption is mandatory if access is provided via another carrier. Since Deutsche Telekom is one of the largest VoIP providers in Germany, it would be nice if Asterisk could be used here. I know the patch from Michael Maier that was mentioned here on the mailing list. I was able to install it successfully together with Asterisk 18.10. Are there any plans to add Mediasec support to Asterisk or to include this patch? Have a nice day. Best regards Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
