Aquileo | Activity for siproxd - SIP proxy/masquerading daemon

Aquileo | Thomas Ries modified ticket #67

Fri, 01 May 2026 13:51:09 -0000

Configure fails on RHEL 10

Aquileo | Thomas Ries posted a comment on ticket #67

Fri, 01 May 2026 13:50:46 -0000

0.8.4 has been released.

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.4/README

Fri, 01 May 2026 13:35:03 -0000

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.4/siproxd-0.8.4.tar.gz

Fri, 01 May 2026 13:34:03 -0000

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.4/siproxd-0.8.4.tar.gz.asc

Fri, 01 May 2026 13:34:03 -0000

Aquileo | Thomas Ries posted a comment on ticket #67

Fri, 24 Apr 2026 16:01:22 -0000

Hello Simon, of course, you may :-)

Aquileo | Simon Matter posted a comment on ticket #67

Fri, 24 Apr 2026 15:52:51 -0000

Hi Thomas, Yes, you are right that fixes it. May I vote for a new release? Thanks, Simon

Aquileo | Thomas Ries posted a comment on ticket #67

Fri, 24 Apr 2026 15:29:00 -0000

Hello Simon, are you using the latest code from Github? According to configure.log this is likely an issue that has been fixed in July 2025 in the master branch (but no release has been made since). -> https://github.com/hb9xar/siproxd/commit/0bb5dd4aac9f5578db664d82f67091bc83c3780d best regards, /Thomas

Aquileo | Simon Matter modified a comment on ticket #67

Fri, 24 Apr 2026 06:47:35 -0000

Hi Thomas, Attached is the first log file. Thanks, Simon

Aquileo | Simon Matter posted a comment on ticket #67

Fri, 24 Apr 2026 06:47:03 -0000

The other log file.

Aquileo | Simon Matter posted a comment on ticket #67

Fri, 24 Apr 2026 06:46:09 -0000

Hi Thomas, Attached are the requested log files. Thanks, Simon

Aquileo | Thomas Ries posted a comment on ticket #67

Thu, 23 Apr 2026 18:28:58 -0000

Hi Simon, can you please provide the full output of ./configure and the resulting "config.log" file? Regards, /Thomas

Aquileo | Simon Matter created ticket #67

Thu, 23 Apr 2026 16:31:33 -0000

Configure fails on RHEL 10

Aquileo | Thomas Ries modified ticket #66

Thu, 12 Mar 2026 20:38:52 -0000

year-2038 issue in plugin_blacklist

Aquileo | Thomas Ries posted a comment on ticket #66

Thu, 12 Mar 2026 20:38:02 -0000

Thank you for reporting this issue. The suggested fix has been implemented with commit 879925a80b5d80f425abeb82d934b1425c7830cd

Aquileo | Thomas Ries modified ticket #66

Mon, 09 Mar 2026 05:45:22 -0000

year-2038 issue in plugin_blacklist

Aquileo | Bernhard M. Wiedemann created ticket #66

Fri, 06 Mar 2026 13:21:41 -0000

year-2038 issue in plugin_blacklist

Aquileo | Tobias Boesch modified a comment on discussion Help

Sun, 11 Jun 2023 18:28:32 -0000

We get multiple phone numbers from the isp. I don't know of they are separate accounts or something else. Are there any experiences on how different "numbers" (SIP accounts?) be "routed" to different routers through siproxd? Checked how my current provider handles this and found out that the SIP account name is the phone number. It is then easy to allocate different numbers to different clients (parties): every party gets its own SIP account.

Aquileo | Tobias Boesch posted a comment on discussion Help

Sun, 11 Jun 2023 18:27:28 -0000

Aquileo | Tobias Boesch modified a comment on discussion Help

Sun, 11 Jun 2023 08:10:32 -0000

Related, but different threads are the two double NAT threads: https://sourceforge.net/p/siproxd/discussion/203640/thread/add33c4b/ and https://sourceforge.net/p/siproxd/discussion/203640/thread/d37b1c51/ The difference here is that the second inner NAT routers we want to use (most likely) have SIP account entry possibilities and therefore hopefully not need a (second) siproxd instance.

Aquileo | Tobias Boesch posted a comment on discussion Help

Sun, 11 Jun 2023 08:09:23 -0000

Regulated threads are double NAT threads: https://sourceforge.net/p/siproxd/discussion/203640/thread/add33c4b/ and https://sourceforge.net/p/siproxd/discussion/203640/thread/d37b1c51/ The difference here is that the second inner NAT routers we want to use (most likely) have SIP account entry possibilities and therefore hopefully not need a (second) siproxd instance.

Aquileo | Tobias Boesch posted a comment on discussion Help

Sun, 11 Jun 2023 07:49:58 -0000

Hello everyone, the network I plan is for a sharing community where four parties live in separate flats and share things and life (a bit of background for better understanding the network requirements). The internet connection should also be shared between these four parties, too (provider knows about this and fully permits this). For this every flat should have a router with NAT (for privacy reasons) and phones connected to the respective router of each flat. All routers are connected to single...

Aquileo | Simon Matter posted a comment on discussion Help

Fri, 16 Dec 2022 07:25:33 -0000

Hi, I was wondering if there is any way to block certain incoming calls? Is there a way I could do this with siproxd? That would be wonderful but I just couldn't find how to do it. Thanks, Simon

Aquileo | David Krugler posted a comment on discussion Help

Wed, 29 Jun 2022 18:11:30 -0000

I ended up adding two more as apparently we could dial out not having to use the local area code.... plugin_regex_desc = prefix outgoing 7 digit number with 1(yyy) plugin_regex_pattern = ^(sips?:)(\+?)(.{7}@) plugin_regex_replace = \1\21yyy\3 plugin_regex_desc = prefix outgoing 7 digits without a leading '+' with a leading '1 (yyy)' plugin_regex_pattern = ^(sips?:)([0-9]{7}@) plugin_regex_replace = \11yyy\2

Aquileo | Thomas Ries posted a comment on discussion Help

Thu, 23 Jun 2022 19:54:29 -0000

1) conditional rewriting using plugin_regex: plugin_regex_desc = prefix outgoing 10 digit numbers with a leading '1' #plugin_regex_pattern = ^(sips?:)(\+?)(.{10}@) plugin_regex_pattern = ^(sips?:)(\+?)([0-9]{10}@) plugin_regex_replace = \1\21\3 or maybe (without '+' prefixed, 10 digit numbers only) plugin_regex_desc = prefix outgoing 10 digit numbers without leading '+' with a leading '1' plugin_regex_pattern = ^(sips?:)([0-9]{10}@) plugin_regex_replace = \11\2 only 10 digit numbers will be processed....

Aquileo | David Krugler posted a comment on discussion Help

Tue, 14 Jun 2022 00:00:30 -0000

So this will be a strange question. We haven't ported our main number yet, but going through our current provider, I am trying to "forward" from our main number to one of the DID numbers, and all I get is dead air. I'm not sure if this is siproxd, or our VOIP provider... Attached is the log... RP is registering the call, but for some reason hits the automated attendant even though I shut it off. I'm guessing RP is getting confused. Flowroute is allowing me to use the user ID when authenticating (so...

Aquileo | Thomas Ries posted a comment on discussion Help

Sat, 11 Jun 2022 08:52:14 -0000

Please try without plugin_prefix being active. Looking at the log, I have the impression that the PBX issues a re-INVITE (directed towards flowroute) for the ongoing call that is being affected by plugin_prefix and causes the call to fail. If plugin_prefix is causing an issue in this setup, there are other ways to prefix outgoing calls with the digit 1 (plugin_regex, allows conditionally rewriting outgoing call targets via regular expressions).

Aquileo | David Krugler posted a comment on discussion Help

Fri, 10 Jun 2022 17:46:15 -0000

currently to dial a number in our office we dial 82025551212 (without the 1) The idea was instead of dialing 812025551212 the user would continue to dial 82025551212 since flowrote requires the 1, but that is only for dialing out.

Aquileo | Thomas Ries posted a comment on discussion Help

Wed, 08 Jun 2022 20:58:13 -0000

ping@invalid -> seems to be used by the provider (34.226.36.32) as dummy "From" address in OPTION requests, probably a "ping" feature to check if the client (your PBX) is alive. NULL@192.168.0.189 -> Response Point does send Contact Headers with no user part (only the hostname part) in INVITE requests. This is technically legal and should not cause any issues with siproxd. In the logs, this missing user part is shown as "NULL" to indicate the missing user. There seems to be an unwanted trigger of...

Aquileo | David Krugler posted a comment on discussion Help

Wed, 08 Jun 2022 17:50:56 -0000

I'm sorry to bug you again, but ran into something trying to get this ready for primetime. We are going to be looking to switch fully to VOIP, and one item is that our main number goes to the receptionist (202-555-1212). For some reason, I don't know if there is some strange static, or some miscommunication with the Proxy, but it dials in, you hear the ring locally for a millisecond, but then RP kicks in the Automated Attendant (from RP's logs) and then goes silent. We are not having an issue with...

Aquileo | David Krugler posted a comment on discussion Help

Wed, 08 Jun 2022 01:09:28 -0000

so if any other "MS Response Point" users are out there and looking at this, I have a site that goes through all the settings.... Many Many Many kudos to Thomas for his help in setting this up, and having some forward thinking code that was able to adapt to what I needed. http://krugler.zapto.org/responsepoint.htm

Aquileo | David Krugler posted a comment on discussion Help

Tue, 07 Jun 2022 21:27:01 -0000

Thomas You the man!!!!!! Thanks for the app. Next time you are in the Twin Cities.....

Aquileo | Thomas Ries posted a comment on discussion Help

Tue, 07 Jun 2022 20:43:38 -0000

My guess would be, you have messed up the siprunk configuration (siptrunk_account): b4: plugin_siptrunk_name = Response Point plugin_siptrunk_account = sip:1yyyyyy8000@us-east-va.sip.flowroute.com <<<<< plugin_siptrunk_numbers_regex = ^+?1yyyyyy985([1-9])$ now: plugin_siptrunk_name = Response Point plugin_siptrunk_account = sip:1yyyyyy9853@us-east-va.sip.flowroute.com <<<<< plugin_siptrunk_numbers_regex = ^\+?1yyyyyy985([1-9])$

Aquileo | David Krugler posted a comment on discussion Help

Tue, 07 Jun 2022 20:31:04 -0000

Thomas, I'm there, and with all this testing, I must have flipped something that I'm not seeing trying to put this into production.... This was working in debug, but when I flipped to "dameaon" it all stopped working

Aquileo | Thomas Ries posted a comment on discussion Help

Tue, 07 Jun 2022 18:17:20 -0000

12:36:16.415 ERROR:plugin_siptrunk.c:168 Regular expression [^+?1yyyyyy985([1-9])$] failed to compile: Invalid preceding regular expression My bad, missed the '\' character in the REGEX (must have gotten lost between brain and keyboard). Actually, the HTML editor of sourceforge does make them disappear. ^\+?1yyyyyy985([1-9])$ That should be the correct REGEX (at least the preview does show it properly).

Aquileo | David Krugler posted a comment on discussion Help

Tue, 07 Jun 2022 17:54:42 -0000

Thomas, I really appreciate your help on this. I modified the trunk number regex to plugin_siptrunk_numbers_regex = ^+?1yyyyyy985([1-9])$ still getting timeout's sent

Aquileo | Thomas Ries posted a comment on discussion Help

Tue, 07 Jun 2022 17:30:44 -0000

Your provider uses two different number formats. In the Request URI a format without leading '+' is used, in the To Header the number format does include the leading '+'. check here: 19:36:56.272 plugin_siptrunk.c:203 Request URI: [1yyyyyy9853] 19:36:56.272 plugin_siptrunk.c:209 To: header: [+1yyyyyy9853] 19:36:56.272 plugin_siptrunk.c:231 plugin_siptrunk: matched trunk on rule 0 [^1yyyyyy985([1-9])$] 19:36:56.273 plugin_siptrunk.c:234 plugin_siptrunk: Trunk [Response Point], Account [sip:1yyyyyy8000@us-east-va.sip.flowroute.com]...

Aquileo | David Krugler posted a comment on discussion Help

Tue, 07 Jun 2022 00:57:33 -0000

Thomas, Thanks again, you assessment was close. with the switching from one network card to two trying to get this to work, i forgot to assign the outbound 0.181 to the xxx.xxx.228.141 in pfsense. So corrected that, but now I think I'm still left with a timeout issue. Same issue where the phone rings, but cannot answer. it appears it is now recieving the ACK packet (time 19:36:58:340) , but doesn't send it to the 189 and instead sends "request Timeout" to 34.226

Aquileo | Thomas Ries posted a comment on discussion Help

Mon, 06 Jun 2022 09:46:41 -0000

The debug log looks fine, I cannot see anything bad in the behavior of siproxd. All outgoing packets look correct. What happens is basically: (You can check RFC3261, page 11 for a graphical representation of the signalling flow) <--- INVITE ---> 100 Trying ---> 180 Ringing phone being picked up ---> 200 OK The required 'ACK' request (completes the call establishmend) from the remote side is never received. Either the outgoing SIP packets never make it to your provider, or the ACK is not sent / sent...

Aquileo | David Krugler posted a comment on discussion Help

Mon, 06 Jun 2022 00:21:13 -0000

Thomas, Thanks, that's doing the trick, partly. The direct dials are now coming in and is ringing, but I can't answer and not sure what is happing. Either the identification is taking too long and it's missing some of the packet, or something is happening in the re-write. Note on line. 419 of the debug log file plugin_siptruck_name = Response Point plugin_siptrunk_account = sip:1yyyyyy8000@us-east-va.sip.flowroute.com plugin_siptrunk_numbers_regex = ^1yyyyyy985([1-9])$

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 05 Jun 2022 10:27:15 -0000

The number you register is 1yyyyyyy8000. This means, at this point only this number is known and handled by siproxd. If you have a whole number broch attached to this account (account being "1yyyyyyy8000"), then you need to pass this information to siproxd using the plugin_siptrunk. load_plugin=plugin_siptrunk.la plugin_siptrunk_name = Response Point plugin_siptrunk_account = sip:1yyyyyyy8000@us-east-va.sip.flowroute.com plugin_siptrunk_numbers_regex = ^$ Waring:...

Aquileo | David Krugler posted a comment on discussion Help

Thu, 02 Jun 2022 23:01:14 -0000

So looking through the code, it looks like the registration table is the key to "routing" my DID's to the system. Again still using the proxy configuration outlined in method 7.5. Since the main VOIP box handles all the calls and routes appropriately, I think I can create a registration table to route all DID's to the main box. Problem is the format of this file isn't clear, and it won't generate a default one for me to build on and I can't find the format in the documentation. Probably why it's...

Aquileo | David Krugler posted a comment on discussion Help

Thu, 02 Jun 2022 04:46:21 -0000

Thomas, thanks for the direction: ok, built new Ubuntu 17.04.6 LTS with libosip2 5.3.0 and siproxd 0.8.3. after commenting out the mask_host and chrootjail, all outbound is working (at least as sudo) the header is getting the public IP as desired, but now I am looking for the setting for all inbound traffic to be routed to my 0.189 box. Obviously the VOIP provider has no idea there is a proxy, and I've looked through the stock variables, even tried host_inbound to set to the 0.189 address, but seems...

Aquileo | Thomas Ries modified a comment on discussion Help

Thu, 26 May 2022 08:54:31 -0000

00:43:03 sock.c:281 received UDP packet from [192.168.0.189:1852] count=422 ---BUFFER DUMP follows--- 52 45 47 49 53 54 45 52 20 73 69 70 3a 75 73 2d REGISTER sip:us- 65 61 73 74 2d 76 61 2e 73 69 70 2e 66 6c 6f 77 east-va.sip.flow 72 6f 75 74 65 2e 63 6f 6d 20 53 49 50 2f 32 2e route.com SIP/2. 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 0..Via: SIP/2.0/ Your Phone requests registration with "us-east-va.sip.flowroute.com", but siproxd (resp. the host running siproxd on) cannot resolve this host...

Aquileo | Thomas Ries posted a comment on discussion Help

Thu, 26 May 2022 08:54:13 -0000

Aquileo | David Krugler posted a comment on discussion Help

Thu, 26 May 2022 01:22:49 -0000

So I have setup a Linux box to act as "proxy" (as outlined in 7.5 of the manual) in order to have our VPN'd phone work properly I have attached my setup files and the log file I got. Best I can make out is in first few lines of the registration, it's sending the local IP (192.168) and I need siproxd to change to xxx.xxx (Addresses changed to protect the innocent) I have also included all I get for configuration on Response Point. I'm guessing there is some misconfiguration or additional setting I...

Aquileo | Henry Pootel posted a comment on discussion Help

Tue, 26 Apr 2022 10:58:22 -0000

Hello. I've configured siproxd to relay all calls to my asterisk. It's work fine. I have two nets: local_phones - 10.1.1.0/24 and to connect an asterisk - 172.16.1.1/30 My phones are in 10.1.1.0/24 network and connect to the siproxd to 10.1.1.1 address. But! If I use a SIP phone with simple router with NAT (which are work fine with my asterisk directly), the siproxd not answer to registration querres. Yes, the SIP phone ask a registration from unroutable network of the router with SIP Message header:...

Aquileo | David Krugler posted a comment on discussion Help

Tue, 26 Apr 2022 03:10:13 -0000

I am looking for some initial settings for an old phone system from Microsoft back in 2010 era days called Response Point. The phone system is a VOIP and we have been using with a POTS gateway for some time. I am looking to move to full VOIP which the system does support. However, because it is behind a firewall (pfsense) and the way that our VOIP provider handles Audio (Incoming calls are routed directly from the carrier the call is originated on), if RP has to send some, I'll say message, because...

Aquileo | Thomas Ries modified ticket #65

Fri, 18 Feb 2022 13:56:45 -0000

support reproducible builds

Aquileo | Thomas Ries posted a comment on ticket #65

Fri, 18 Feb 2022 13:56:16 -0000

https://github.com/hb9xar/siproxd/commit/4750bea4ffedb4543a404dafc979c2b16b53e523 implements support for SOURCE_DATE_EPOCH

Aquileo | Bernhard M. Wiedemann posted a comment on ticket #65

Fri, 18 Feb 2022 07:53:40 -0000

Thanks. I propose this addition: https://github.com/hb9xar/siproxd/pull/3

Aquileo | Thomas Ries modified ticket #43

Fri, 18 Feb 2022 07:07:54 -0000

PATCH to 0.7.2 need host_inbound for virtual lan ip?

Aquileo | Thomas Ries modified ticket #59

Fri, 18 Feb 2022 07:06:06 -0000

Create UA RTP port mapping dynamically using UPnP/NAT-PCP to remove the need for RTP proxy

Aquileo | Thomas Ries modified ticket #61

Fri, 18 Feb 2022 07:03:20 -0000

Outgoing SIP Calls terminated after 30 seconds

Aquileo | Thomas Ries posted a comment on ticket #61

Fri, 18 Feb 2022 07:03:20 -0000

being closed due to inactivity

Aquileo | Thomas Ries modified ticket #65

Fri, 18 Feb 2022 06:59:51 -0000

support reproducible builds

Aquileo | Thomas Ries posted a comment on ticket #65

Fri, 18 Feb 2022 06:59:34 -0000

being worked on, check https://github.com/hb9xar/siproxd/commit/f47a3e0f0bcd04b425fe31b547b99c19f0a7d9f7

Aquileo | Bernhard M. Wiedemann created ticket #65

Tue, 15 Feb 2022 07:24:12 -0000

support reproducible builds

Aquileo | Alexander Dahl created ticket #64

Tue, 09 Mar 2021 15:31:57 -0000

fails at compile time if sqlite3 is missing

Aquileo | Jean-Jacques Sarton created ticket #63

Sun, 14 Feb 2021 17:14:00 -0000

plungin_stun segment violation

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Thu, 11 Feb 2021 08:10:20 -0000

Tested with gcc 10.2 and gcc 7.4, work well with both.

Aquileo | Thomas Ries posted a comment on discussion Help

Thu, 11 Feb 2021 07:00:14 -0000

git master should now build without warnings (gcc 9.3.0)

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Mon, 08 Feb 2021 19:24:28 -0000

I had made tests with the correct setting at this time this what not okay. I have corrected my mistake and now the test what successful. Thanks. By the way the beta version produce further warnings, see attached file. Instead of using sprintf() yo may use snprintf(). The declaration of ident produce also warning, sing that what nice in old time.

Aquileo | Thomas Ries posted a comment on discussion Help

Mon, 08 Feb 2021 17:07:26 -0000

Looking at the debug log, it seems that you have mixed up the inbound and outbound networks. the local UA (phone) must be located in the inbound network the Registrar (PBX) must be located in the outbound network This of course will mess up everything, masquerading stuff with the wrong IP addresses, etc. Also, that's the reason why the fritzbox does send RTP traffic to the 10.x.x.x side of siproxd and not to the 192.168.x.x address of the siproxd host.

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Mon, 08 Feb 2021 14:27:40 -0000

I have run siproxd on an other system, which has no NAT helper and this worked. This don't solve my problem but the reason is more or less found.

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Mon, 08 Feb 2021 12:05:27 -0000

No NAT helper is involved on the system with Linphone. On the Server NAS the modules are loaded but the rules are only for docker instances and shall not interfere with siproxd. For the tunnel there are also no iptables rules containing problematic thinks.

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 07 Feb 2021 21:24:41 -0000

OF course, you are right. My bad.

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Sun, 07 Feb 2021 21:11:38 -0000

I know, but via->port[5-1] will short the string if the port number is greater as 9999. The code may be eliminated and will only be effective if there is an error in the code or the called function above. Am 07.02.21 um 19:59 schrieb Thomas Ries: No. snprintf does include the terminating \0, also if the size limit is triggered. The limit is /including/ the terminating \0 (-> 6 = 5 digits + terminating \0). compile warning https://sourceforge.net/p/siproxd/discussion/203640/thread/e783812b9f/?limit=25#372f...

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 07 Feb 2021 18:59:32 -0000

No. snprintf does include the terminating \0, also if the size limit is triggered. The limit is including the terminating \0 (-> 6 = 5 digits + terminating \0).

Aquileo | Jean-Jacques Sarton modified a comment on discussion Help

Sun, 07 Feb 2021 17:27:58 -0000

What is with line 159 should it not been: via->port[PORTSTRING_SIZE-1] ='\0'; or via->port[5] ='\0';

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Sun, 07 Feb 2021 17:20:20 -0000

What is with line 159 should it not been: via->port[PORTSTRING_SIZE-1] ='\0';

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 07 Feb 2021 17:08:02 -0000

First, make really sure that no NAT helpers are involved on the system where siproxd is running. 95% of RTP related problems originate by NAT helpers (like SIP conntrack kernel modules). Second, to really see what is going on in your case, a debug log is needed (including the REGISTER and INVITE sequence, up to the failure). See 6.1. Problem Reporting

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 07 Feb 2021 16:59:57 -0000

Should be fixed in git master by now.

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Sat, 06 Feb 2021 16:42:17 -0000

Hello, I have the following scenario Siproxd is installed on a NAS system which is connected to the LAN and work also as VPN server. LAN IP: 192.178.178.2 VPN IP: 10.10.10.2 The Router is a FritzBox and allow telephonie LAN IP: 192.178.178.1 A Smartphone or a PC ist connected to the internet and shall allow calls through the FritzBox WAN IP: a.b.c.d VPN IP: 10.10.10.3 If I make a Call via the Device attached to the WAN with Linphone the SIP messages are forwarded as expected. For the RTP stream the...

Aquileo | Jean-Jacques Sarton posted a comment on discussion Help

Sat, 06 Feb 2021 11:54:05 -0000

I have the following warning while compiling siproxd-0.8.4dev-9 plugin_fix_bogus_via.c:158:7: note: 'snprintf' output between 2 and 6 bytes into a destination of size 5 snprintf(via->port, 5, "%u", ntohs(ticket->from.sin_port)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

Aquileo | Thomas Ries posted a comment on discussion Help

Mon, 25 Jan 2021 16:58:12 -0000

The debug log fragment is too short. To have a chance to properly analyze request processing, the log must be taken with debuglevel=-1 and it should include at least: - Registration - Call Setup, up to the point where the malfunction is observed The above fragment only seems to cover part of the BYE sequence (this is just a guess), so no change to figure out what is really going on there. If you are concerned about publishing confidential information with your log you may send the log directly to...

Aquileo | Ricardo posted a comment on discussion Help

Thu, 21 Jan 2021 14:38:02 -0000

Dear guys, We have siproxd running between 2 networks. The sip-client is in a separated (not routed), and we have an asterisk in another network. Between these networks is siproxd running. We have to user transparent mode. Everything what uses SIP protocoll works (client is connected to asterisk). when I initiat a call from asterisk to client, the client can answer the call. But both can't hear each other. I can see RTP packets on both side: Asterisk (10.0.010) --> Router with siproxd (ext 10.0.01...

Aquileo | mterwoord posted a comment on discussion Help

Mon, 18 Jan 2021 09:41:18 -0000

Hi Thomas, Thanks for replying. I just installed 3 Yeastar S50 PBX-es, but turns out the S series pretty much is legacy, as they don't do IPv6. I'm now looking for a way to have mobile extensions (cell phones) connect via IPv6 to the pbx's. With kind regards, Matthijs ter Woord Op zo 17 jan. 2021 om 14:11 schreef Thomas Ries

Aquileo | Thomas Ries posted a comment on discussion Help

Sun, 17 Jan 2021 13:11:49 -0000

Hi, siproxd does not have support for IPv6. It is not possible to proxy IPv6 SIP traffic. Regards, /Thomas

Aquileo | mterwoord posted a comment on discussion Help

Thu, 14 Jan 2021 08:04:50 -0000

Hi, I have a PBX which only does IPv4. Is it possible to use siproxd to proxy IPv6 traffic to the PBX? Thanks!

Aquileo | Thomas Ries posted a comment on discussion Help

Fri, 08 Jan 2021 14:53:53 -0000

Yes, the plugin_siptrunk will help you with this. # Plugin_siptrunk # # Plugin to handle SIP Trunks where using *one* single SIP account a # whole number block is routed. This means an incoming INVITE does carry # the target number (in SIP URI or To: header field) but does not really # carry any clear indications to which account it belongs to. # Thus, we need some help - a mapping of the number blocks used in a SIP # trunk and the corresponding SIP account (as used during REGISTER) # # ..._name:...

Aquileo | Matthias Hagen posted a comment on discussion Help

Fri, 08 Jan 2021 09:39:31 -0000

I need Help with my Siproxd. I had an OpenSense Firewall and behind the Firewall an telephone system, from t he company Agfeo. I use one Sip-Trunk Account. I use 10 phones on my telephone system. I will that you can call all thes phones directly from outside. At the moment you can only call the main-number (0557783310). If you call this number everthing works perfect. If you call a phone directly, for example 05577 83310 12, the call didn´t work. I search a lot in the Internet. I think what I need...

Aquileo | Stef posted a comment on discussion Help

Tue, 29 Dec 2020 10:15:07 -0000

Hi Thomas, I confirm that the latest bleedingedge snapshot does the trick! Record-Route headers are detected & removed as expected. INVITE works properly now, but I have some other issues I must resolve - I am not sure yet if the latest snapshot is the source of the problem. Thank you very much!

Aquileo | Thomas Ries posted a comment on discussion Help

Mon, 28 Dec 2020 10:28:31 -0000

Hi Stef, Can you try the latest bleedingedge snapshot or the master branch of the git repo? /Thomas

Aquileo | Stef posted a comment on discussion Help

Sat, 26 Dec 2020 13:48:51 -0000

Hi Thomas, the info you requested is following. I have striped down the information to the absolute necessary, plus I removed IPs,ports, etc. for privacy. configuration partial snapshot #plugin_stripheader_remove = Allow plugin_stripheader_remove = User-Agent plugin_stripheader_remove = Record-Route # remove only a particular value from a header (no spaces allowed) #plugin_stripheader_remove = Supported:100rel debug log partial snapshot 21:19:43.183 plugin_stripheader.c:106 plugin_stripheader: looking...

Aquileo | Thomas Ries posted a comment on discussion Help

Fri, 25 Dec 2020 20:21:31 -0000

Hi, How does your configuration look like (the part for plugin_stripheader)? Can you provide a debug log covering the reception, processing and transmission of such a SIP packet? Please note that plugin_stripheader is called before all the processing is done, so "Record-Route" headers added by siproxd itself (as part of the processing) will still be present. /Thomas

Aquileo | Stef posted a comment on discussion Help

Tue, 15 Dec 2020 12:27:42 -0000

Hello all, I am trying to use the stripheader plugin in order to remove some "Record-Route" entries in INVITE headers, that cause trouble in my device. Although the plugin works as expected in removing "user-agent" entries for example, the "Record-Route" entries are just ignored. Is there a limitation in which entries the stripheader plugin can remove? Is there any other way that I may remove or alter the "Record-Route" entries?

Aquileo | Thomas Ries created a blog post

Mon, 24 Aug 2020 20:04:00 -0000

New Siproxd Release 0.8.3

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.3/README

Mon, 24 Aug 2020 19:55:03 -0000

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.3/siproxd-0.8.3.tar.gz

Mon, 24 Aug 2020 19:54:03 -0000

Aquileo | siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.3/siproxd-0.8.3.tar.gz.asc

Mon, 24 Aug 2020 19:54:03 -0000

Aquileo | Thomas Ries modified ticket #62

Sun, 26 Jul 2020 09:24:34 -0000

siproxd and gcc-10: multiple definition of `configuration' error

Aquileo | Thomas Ries posted a comment on ticket #62

Sun, 26 Jul 2020 09:23:40 -0000

Hello Seb, Thank you for bringing this to my attention. Yes, the proposed patch is OK. This will be fixed in the next released version of siproxd. Best regards, /Thomas

Aquileo | micmac1 created ticket #62

Sun, 26 Jul 2020 08:29:38 -0000

siproxd and gcc-10: multiple definition of `configuration' error

Aquileo | Thomas Ries posted a comment on discussion Help

Sat, 15 Feb 2020 09:38:09 -0000

As stated above, if you configure your UA properly, those two headers will be sent correctly by the UA itself.

Aquileo | Phillip posted a comment on discussion Help

Fri, 14 Feb 2020 22:42:18 -0000

That's a shame, application works well, except for those two headers.

Aquileo | Thomas Ries posted a comment on discussion Help

Fri, 14 Feb 2020 20:51:22 -0000

This is a basic design decision that has been made in the very beginning. Siproxd behaves like a proxy It alters header required for NAT traversal It shall not have any knowledge of user credentials (which would be required in a back2back design) There shall be no configuration required in siproxd for individual UAs traversing siproxd. In a back2back design, there would be the requirement to know about the UAs as well as the mapping of the external to internal SIP accounts. This design has its specific...

Aquileo | Phillip posted a comment on discussion Help

Fri, 14 Feb 2020 19:49:39 -0000

Thx Thomas, i fugured by the name (Siproxd) is a proxy and not a B2BUA :) In any event, a true proxy doesn't alter headers, it only adds it's own via header, record route, decreases the max forward count, etc. Since Siproxd is behaving more like a hybrid proxy, i thought since it already alters certain headers, why not one more?

Aquileo | Thomas Ries posted a comment on discussion Help

Wed, 12 Feb 2020 17:49:30 -0000

Siproxd acts as a proxy. This means that on your User-Agent (phone) you configure the account with the credentials as provided by your registrar. If your local client (phone) is configured correctly that way, the From header and SIP URIs.

Aquileo | Phillip posted a comment on discussion Help

Tue, 11 Feb 2020 22:57:46 -0000

This file was automatically generated by the pfSense package management system. if_inbound = em1 if_outbound = em0 sip_listen_port = 5060 daemonize = 1 silence_log = 0 user = nobody chrootjail = /var/siproxd/ registration_file = siproxd_registrations autosave_registrations = 10 pid_file = siproxd.pid rtp_proxy_enable = 1 rtp_port_low = 7070 rtp_port_high = 7079 rtp_timeout = 300 default_expires = 60 debug_level = -1 outbound_domain_name = quovim.voip outbound_domain_host = sip.quovim.voip outbound_domain_port...