Activity for siproxd - SIP proxy/masquerading daemon
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Tobias Boesch modified a comment on discussion Help
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.
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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).
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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.
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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.
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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...
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Simon Matter posted a comment on discussion Help
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
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David Krugler posted a comment on discussion Help
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
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Thomas Ries posted a comment on discussion Help
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....
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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...
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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).
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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.
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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...
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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...
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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
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Thomas You the man!!!!!! Thanks for the app. Next time you are in the Twin Cities.....
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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])$
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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
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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).
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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
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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])$] 19:36:56.273 plugin_siptrunk.c:234 plugin_siptrunk: Trunk [Response Point], Account [sip:1yyyyyy8000@us-east-va.sip.flowroute.com]...
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Thomas, Thanks again, you assessment was close. with the switching from one network card to two trying to get this to work, i forgot to assign the outbound 0.181 to the xxx.xxx.228.141 in pfsense. So corrected that, but now I think I'm still left with a timeout issue. Same issue where the phone rings, but cannot answer. it appears it is now recieving the ACK packet (time 19:36:58:340) , but doesn't send it to the 189 and instead sends "request Timeout" to 34.226
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The debug log looks fine, I cannot see anything bad in the behavior of siproxd. All outgoing packets look correct. What happens is basically: (You can check RFC3261, page 11 for a graphical representation of the signalling flow) <--- INVITE ---> 100 Trying ---> 180 Ringing phone being picked up ---> 200 OK The required 'ACK' request (completes the call establishmend) from the remote side is never received. Either the outgoing SIP packets never make it to your provider, or the ACK is not sent / sent...
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Thomas, Thanks, that's doing the trick, partly. The direct dials are now coming in and is ringing, but I can't answer and not sure what is happing. Either the identification is taking too long and it's missing some of the packet, or something is happening in the re-write. Note on line. 419 of the debug log file plugin_siptruck_name = Response Point plugin_siptrunk_account = sip:1yyyyyy8000@us-east-va.sip.flowroute.com plugin_siptrunk_numbers_regex = ^1yyyyyy985([1-9])$
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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 = ^<REGEX to match the whole number block>$ Waring:...
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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...
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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...
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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...
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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...
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David Krugler posted a comment on discussion Help
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...
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Henry Pootel posted a comment on discussion Help
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:...
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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...
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support reproducible builds
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https://github.com/hb9xar/siproxd/commit/4750bea4ffedb4543a404dafc979c2b16b53e523 implements support for SOURCE_DATE_EPOCH
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Bernhard M. Wiedemann posted a comment on ticket #65
Thanks. I propose this addition: https://github.com/hb9xar/siproxd/pull/3
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PATCH to 0.7.2 need host_inbound for virtual lan ip?
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Create UA RTP port mapping dynamically using UPnP/NAT-PCP to remove the need for RTP proxy
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Outgoing SIP Calls terminated after 30 seconds
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being closed due to inactivity
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support reproducible builds
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being worked on, check https://github.com/hb9xar/siproxd/commit/f47a3e0f0bcd04b425fe31b547b99c19f0a7d9f7
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support reproducible builds
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Alexander Dahl created ticket #64
fails at compile time if sqlite3 is missing
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Jean-Jacques Sarton created ticket #63
plungin_stun segment violation
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Tested with gcc 10.2 and gcc 7.4, work well with both.
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git master should now build without warnings (gcc 9.3.0)
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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.
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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.
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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.
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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.
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OF course, you are right. My bad.
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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...
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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).
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What is with line 159 should it not been: via->port[PORTSTRING_SIZE-1] ='\0'; or via->port[5] ='\0';
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What is with line 159 should it not been: via->port[PORTSTRING_SIZE-1] ='\0';
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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
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Should be fixed in git master by now.
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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...
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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)); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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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...
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Ricardo posted a comment on discussion Help
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...
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mterwoord posted a comment on discussion Help
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 <tries@users.sourceforge.net : Hi, siproxd does not have support for IPv6. It is not possible to proxy IPv6 SIP traffic. Regards, /Thomas IPv6 to IPv4 translation https://sourceforge.net/p/siproxd/discussion/203640/thread/a844d223fc/?limit=25#e15d...
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Hi, siproxd does not have support for IPv6. It is not possible to proxy IPv6 SIP traffic. Regards, /Thomas
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Hi, I have a PBX which only does IPv4. Is it possible to use siproxd to proxy IPv6 traffic to the PBX? Thanks!
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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:...
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Matthias Hagen posted a comment on discussion Help
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...
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Stef posted a comment on discussion Help
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!
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Hi Stef, Can you try the latest bleedingedge snapshot or the master branch of the git repo? /Thomas
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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...
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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
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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?
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New Siproxd Release 0.8.3
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siproxd - SIP proxy/masquerading daemon released /siproxd/0.8.3/README
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siproxd and gcc-10: multiple definition of `configuration' error
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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
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micmac1 created ticket #62
siproxd and gcc-10: multiple definition of `configuration' error
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As stated above, if you configure your UA properly, those two headers will be sent correctly by the UA itself.
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Phillip posted a comment on discussion Help
That's a shame, application works well, except for those two headers.
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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...
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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?
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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.
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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...
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Hello all, I've searched through this forum and found many interesting topics that somewhat covered my problem. Basically, i'm evaluating PFsense with the siproxd module runnning. The issue i'm facing is with the sip invite URI and the sip to URI. It seems siproxd doesn't change these value to reflect the IP of the Sip trunk provider. The contact heder is changed, the SDP is correct (WAN IP with defined rtp port in pfsense), but the other two headers mentioned above are left intact. Is there any...
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Log at around 16:19:46: To me it looks like either: phone does not receive the 200 OK send by siproxd or phone does not send (or siproxd does not receive) the ACK This causes repetitions of the 200 OK (from the registrar). Same later in the log: 16:20:03 outgoing INVITE 16:20:06 incoming "200 OK" that is not ack'd by the phone with an ACK repetitions "200 OK" from registrar 16:20:18 BYE from regsistrar (Reason: SIP;cause=408;text="Request Timeout (1:2)" You have to figure out (network trace on the...
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Stephan posted a comment on ticket #61
I´ve appended the Log. It contains everything from registering, a failed try of a call (I was to fast), a working Inbound call and the failing outbound call. Von: Thomas Riestries@users.sourceforge.net Gesendet: Sonntag, 5. Januar 2020 17:00 An: [siproxd:bugs] 61@bugs.siproxd.p.re.sourceforge.net Betreff: [siproxd:bugs] #61 Outgoing SIP Calls terminated after 30 seconds Please contact me directly via my SF email address: tries at users .sourceforge .net [bugs:#61]https://eur05.safelinks.protection.outlook.com/?url=https%3A%2F%2Fsourceforge.net%2Fp%2Fsiproxd%2Fbugs%2F61%2F&data=02%7C01%7C%7Cb226a1cccc624ffed25d08d791f866b5%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637138368362653642&sdata=BLumB24aRvvSrlO9T8uFRJJ02doMRgO%2FDS%2FHUI0fd8A%3D&reserved=0...
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Thanks. I´ve send you a mail with the log.
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Please contact me directly via my SF email address: tries at users .sourceforge .net
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Thanks for the quick answer. I´ve generated the log successfully, but how can I send you the log directly? I don´t want my personal phone number be available to everyone and searching a 2 MB log file including SIP packets takes a lot of time.
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Create a fully detailed debug log (debug_level=-1) - this includes sufficient information to analyze the SIP traffic, internal doings of siproxd and usually allows to diagnose a problem quickly.
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How can I create a debug log only with SIP and RTP entries? The config and siproxd´s documentation only show the required options, but not if multiple values are possible.
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Outgoing SIP Calls terminated after 30 seconds
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TCP fragment reassembly fails
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TCP fragment reassembly fails
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Thank you. This fix will be included in the next release as well as into the snapshots.
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Frank Schmirler created ticket #60
TCP fragment reassembly fails
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Toerless Eckert posted a comment on discussion Help
On Sat, Mar 09, 2019 at 10:09:26AM -0000, Thomas Ries wrote: Siproxd is designed to cover the typical case where VoIP equipment in a local network using RFC1918 address space (private IP adrdesses) needs to communicate with a Registrar in the public Internet. This is the typical home- and small office setup. Understood. Your use case is completely different. I'm not saying it can't be done, but it ist not a scenario that is considered supported by siproxd. Also I believe that it really depends on...
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More information about NAT can be found here: https://en.wikipedia.org/wiki/Network_address_translation (see especially the section "Applications affected by NAT") Siproxd is designed to cover the typical case where VoIP equipment in a local network using RFC1918 address space (private IP adrdesses) needs to communicate with a Registrar in the public Internet. This is the typical home- and small office setup. Your use case is completely different. I'm not saying it can't be done, but it ist not a...
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On Fri, Mar 08, 2019 at 06:35:06AM -0000, Thomas Ries wrote: Hi, No this will not work. Siproxd is desgined to help with NATs where the Client (phone) is in a private network segment and the PBX / SIP provider is in the publicc Internet. you are basically trying to do the reverse. Thanks for the quick answer!! Too bad ;-(( Any suggestion what my most simple option would be to solve this issue ? Also: Just for curiosity and only if there is any chance for a non-SIP person like myself to understand...
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Hi, No this will not work. Siproxd is desgined to help with NATs where the Client (phone) is in a private network segment and the PBX / SIP provider is in the publicc Internet. you are basically trying to do the reverse. /Thomas
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