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From: Ihor O. <igo...@gm...> - 2024-08-26 19:15:57
|
Hello, I have SIPp 3.7.2 with TLS support (Dockerfile) as a part of my project https://github.com/igorolhovskiy/volts/blob/main/build/Dockerfile.sipp How it's called can be found in a following python code parts: https://github.com/igorolhovskiy/volts/blob/main/build/src/sipp/sipp.py#L41 https://github.com/igorolhovskiy/volts/blob/main/build/src/sipp/sipp.py#L62 Hope, this can give a hint. Le 26/08/2024 à 12:10, Jafar Sarif a écrit : > > Dear SIPp Community, > > I hope this message finds you well. > > I am currently facing an issue while running a SIPp XML scenario over > TLS. Upon execution, I receive the following error message: > TLS_init_context: SSL_CTX_use_certificate_file failed: > error:02001002:system library:fopen:No such file or directory. > > It seems that SIPp is unable to locate the required certificate file > during the TLS initialization process. I have ensured that OpenSSL is > properly installed on the system and that all necessary header files > are present. However, despite these checks, the issue persists. > > Could you kindly assist me in identifying the cause of this error and > provide guidance on resolving the issue? Specifically, I would > appreciate your advice on how to correctly configure SIPp to load the > required certificate file for TLS communication. > > > On Sat, Aug 24, 2024, 3:42 AM Jafar Sarif <sap...@gm...> wrote: > > I will try installing the new version. > Thanks for your help. > > > On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...> > wrote: > > Hi, > I am using SIPp 3.3V which uses makefile for its build system. > > > On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi > <may...@gm...> wrote: > > > > On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif > <sap...@gm...> wrote: > > I am writing to seek assistance with compiling SIPp to > enable TLS support. I have successfully installed SIPp > but encountered difficulties when attempting to > compile it with TLS support. > > *Issue* *Details:* > > Current Installation: I have SIPp version 3.3 > installed, but it was compiled without TLS support. > > OpenSSL Installation: I have confirmed that OpenSSL is > installed and working correctly on my system. The > OpenSSL libraries and headers are available, and I > have verified their installation with the following > commands: > > 1. *openssl* *version* shows: > OpenSSL 1.1.1k FIPS 25 Mar 2021 > > 2. The headers are present in /*usr/include/openss* > > 3. The libraries are located in /*usr/lib*, including > *libssl* and *libcrypto* > > > *Compilation Attempt:* > I set the environment variables for OpenSSL: > > *export CFLAGS="-I/usr/include/openssl"* > * > * > *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* > > After setting these variables, I ran *make* *clean* > followed by *make* in the SIPp directory, but the > issue persists. > > *Error Message:* When I attempt to use TLS with SIPp, > it gives the message: “To use a TLS transport you must > compile SIPp with OpenSSL.” > > Request: > > I would appreciate any guidance or steps to ensure > that SIPp is compiled correctly with TLS support. If > there are specific steps or configurations needed to > resolve this issue, please let me know. > > Thank you for your assistance. > > > Hi, > I think you skipped the instruction from > https://github.com/SIPp/sipp?tab=readme-ov-file#building > There is no need to export such variables. > SIPp is built using cmake so you should do something like: > > cmake . -DUSE_SSL=1 > make > make install > > > > > > > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
From: Jafar S. <sap...@gm...> - 2024-08-26 10:10:58
|
Dear SIPp Community, I hope this message finds you well. I am currently facing an issue while running a SIPp XML scenario over TLS. Upon execution, I receive the following error message: TLS_init_context: SSL_CTX_use_certificate_file failed: error:02001002:system library:fopen:No such file or directory. It seems that SIPp is unable to locate the required certificate file during the TLS initialization process. I have ensured that OpenSSL is properly installed on the system and that all necessary header files are present. However, despite these checks, the issue persists. Could you kindly assist me in identifying the cause of this error and provide guidance on resolving the issue? Specifically, I would appreciate your advice on how to correctly configure SIPp to load the required certificate file for TLS communication. On Sat, Aug 24, 2024, 3:42 AM Jafar Sarif <sap...@gm...> wrote: > I will try installing the new version. > Thanks for your help. > > On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...> wrote: > >> Hi, >> I am using SIPp 3.3V which uses makefile for its build system. >> >> On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...> >> wrote: >> >>> >>> >>> On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> >>> wrote: >>> >>>> I am writing to seek assistance with compiling SIPp to enable TLS >>>> support. I have successfully installed SIPp but encountered difficulties >>>> when attempting to compile it with TLS support. >>>> >>>> *Issue* *Details:* >>>> >>>> Current Installation: I have SIPp version 3.3 installed, but it was >>>> compiled without TLS support. >>>> >>>> OpenSSL Installation: I have confirmed that OpenSSL is installed and >>>> working correctly on my system. The OpenSSL libraries and headers are >>>> available, and I have verified their installation with the following >>>> commands: >>>> >>>> 1. *openssl* *version* shows: >>>> OpenSSL 1.1.1k FIPS 25 Mar 2021 >>>> >>>> 2. The headers are present in /*usr/include/openss* >>>> >>>> 3. The libraries are located in /*usr/lib*, including *libssl* and >>>> *libcrypto* >>>> >>>> >>>> *Compilation Attempt:* >>>> I set the environment variables for OpenSSL: >>>> >>>> *export CFLAGS="-I/usr/include/openssl"* >>>> >>>> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* >>>> >>>> After setting these variables, I ran *make* *clean* followed by *make* >>>> in the SIPp directory, but the issue persists. >>>> >>>> *Error Message:* When I attempt to use TLS with SIPp, it gives the >>>> message: “To use a TLS transport you must compile SIPp with OpenSSL.” >>>> >>>> Request: >>>> >>>> I would appreciate any guidance or steps to ensure that SIPp is >>>> compiled correctly with TLS support. If there are specific steps or >>>> configurations needed to resolve this issue, please let me know. >>>> >>>> Thank you for your assistance. >>>> >>>> >>> Hi, >>> I think you skipped the instruction from >>> https://github.com/SIPp/sipp?tab=readme-ov-file#building >>> There is no need to export such variables. >>> SIPp is built using cmake so you should do something like: >>> >>> cmake . -DUSE_SSL=1 >>> make >>> make install >>> >>> >>> >>> >>> >>> >> |
From: Jafar S. <sap...@gm...> - 2024-08-23 22:13:14
|
I will try installing the new version. Thanks for your help. On Sat, Aug 24, 2024, 3:40 AM Jafar Sarif <sap...@gm...> wrote: > Hi, > I am using SIPp 3.3V which uses makefile for its build system. > > On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...> > wrote: > >> >> >> On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote: >> >>> I am writing to seek assistance with compiling SIPp to enable TLS >>> support. I have successfully installed SIPp but encountered difficulties >>> when attempting to compile it with TLS support. >>> >>> *Issue* *Details:* >>> >>> Current Installation: I have SIPp version 3.3 installed, but it was >>> compiled without TLS support. >>> >>> OpenSSL Installation: I have confirmed that OpenSSL is installed and >>> working correctly on my system. The OpenSSL libraries and headers are >>> available, and I have verified their installation with the following >>> commands: >>> >>> 1. *openssl* *version* shows: >>> OpenSSL 1.1.1k FIPS 25 Mar 2021 >>> >>> 2. The headers are present in /*usr/include/openss* >>> >>> 3. The libraries are located in /*usr/lib*, including *libssl* and >>> *libcrypto* >>> >>> >>> *Compilation Attempt:* >>> I set the environment variables for OpenSSL: >>> >>> *export CFLAGS="-I/usr/include/openssl"* >>> >>> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* >>> >>> After setting these variables, I ran *make* *clean* followed by *make* >>> in the SIPp directory, but the issue persists. >>> >>> *Error Message:* When I attempt to use TLS with SIPp, it gives the >>> message: “To use a TLS transport you must compile SIPp with OpenSSL.” >>> >>> Request: >>> >>> I would appreciate any guidance or steps to ensure that SIPp is compiled >>> correctly with TLS support. If there are specific steps or configurations >>> needed to resolve this issue, please let me know. >>> >>> Thank you for your assistance. >>> >>> >> Hi, >> I think you skipped the instruction from >> https://github.com/SIPp/sipp?tab=readme-ov-file#building >> There is no need to export such variables. >> SIPp is built using cmake so you should do something like: >> >> cmake . -DUSE_SSL=1 >> make >> make install >> >> >> >> >> >> > |
From: Jafar S. <sap...@gm...> - 2024-08-23 22:10:39
|
Hi, I am using SIPp 3.3V which uses makefile for its build system. On Sat, Aug 24, 2024, 3:25 AM mayamatakeshi <may...@gm...> wrote: > > > On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote: > >> I am writing to seek assistance with compiling SIPp to enable TLS >> support. I have successfully installed SIPp but encountered difficulties >> when attempting to compile it with TLS support. >> >> *Issue* *Details:* >> >> Current Installation: I have SIPp version 3.3 installed, but it was >> compiled without TLS support. >> >> OpenSSL Installation: I have confirmed that OpenSSL is installed and >> working correctly on my system. The OpenSSL libraries and headers are >> available, and I have verified their installation with the following >> commands: >> >> 1. *openssl* *version* shows: >> OpenSSL 1.1.1k FIPS 25 Mar 2021 >> >> 2. The headers are present in /*usr/include/openss* >> >> 3. The libraries are located in /*usr/lib*, including *libssl* and >> *libcrypto* >> >> >> *Compilation Attempt:* >> I set the environment variables for OpenSSL: >> >> *export CFLAGS="-I/usr/include/openssl"* >> >> *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* >> >> After setting these variables, I ran *make* *clean* followed by *make* >> in the SIPp directory, but the issue persists. >> >> *Error Message:* When I attempt to use TLS with SIPp, it gives the >> message: “To use a TLS transport you must compile SIPp with OpenSSL.” >> >> Request: >> >> I would appreciate any guidance or steps to ensure that SIPp is compiled >> correctly with TLS support. If there are specific steps or configurations >> needed to resolve this issue, please let me know. >> >> Thank you for your assistance. >> >> > Hi, > I think you skipped the instruction from > https://github.com/SIPp/sipp?tab=readme-ov-file#building > There is no need to export such variables. > SIPp is built using cmake so you should do something like: > > cmake . -DUSE_SSL=1 > make > make install > > > > > > |
From: mayamatakeshi <may...@gm...> - 2024-08-23 21:55:16
|
On Sat, Aug 24, 2024 at 6:44 AM Jafar Sarif <sap...@gm...> wrote: > I am writing to seek assistance with compiling SIPp to enable TLS support. > I have successfully installed SIPp but encountered difficulties when > attempting to compile it with TLS support. > > *Issue* *Details:* > > Current Installation: I have SIPp version 3.3 installed, but it was > compiled without TLS support. > > OpenSSL Installation: I have confirmed that OpenSSL is installed and > working correctly on my system. The OpenSSL libraries and headers are > available, and I have verified their installation with the following > commands: > > 1. *openssl* *version* shows: > OpenSSL 1.1.1k FIPS 25 Mar 2021 > > 2. The headers are present in /*usr/include/openss* > > 3. The libraries are located in /*usr/lib*, including *libssl* and > *libcrypto* > > > *Compilation Attempt:* > I set the environment variables for OpenSSL: > > *export CFLAGS="-I/usr/include/openssl"* > > *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* > > After setting these variables, I ran *make* *clean* followed by *make* in > the SIPp directory, but the issue persists. > > *Error Message:* When I attempt to use TLS with SIPp, it gives the > message: “To use a TLS transport you must compile SIPp with OpenSSL.” > > Request: > > I would appreciate any guidance or steps to ensure that SIPp is compiled > correctly with TLS support. If there are specific steps or configurations > needed to resolve this issue, please let me know. > > Thank you for your assistance. > > Hi, I think you skipped the instruction from https://github.com/SIPp/sipp?tab=readme-ov-file#building There is no need to export such variables. SIPp is built using cmake so you should do something like: cmake . -DUSE_SSL=1 make make install |
From: Jafar S. <sap...@gm...> - 2024-08-23 21:41:57
|
I am writing to seek assistance with compiling SIPp to enable TLS support. I have successfully installed SIPp but encountered difficulties when attempting to compile it with TLS support. *Issue* *Details:* Current Installation: I have SIPp version 3.3 installed, but it was compiled without TLS support. OpenSSL Installation: I have confirmed that OpenSSL is installed and working correctly on my system. The OpenSSL libraries and headers are available, and I have verified their installation with the following commands: 1. *openssl* *version* shows: OpenSSL 1.1.1k FIPS 25 Mar 2021 2. The headers are present in /*usr/include/openss* 3. The libraries are located in /*usr/lib*, including *libssl* and *libcrypto* *Compilation Attempt:* I set the environment variables for OpenSSL: *export CFLAGS="-I/usr/include/openssl"* *export LDFLAGS="-L/usr/lib -lssl -lcrypto"* After setting these variables, I ran *make* *clean* followed by *make* in the SIPp directory, but the issue persists. *Error Message:* When I attempt to use TLS with SIPp, it gives the message: “To use a TLS transport you must compile SIPp with OpenSSL.” Request: I would appreciate any guidance or steps to ensure that SIPp is compiled correctly with TLS support. If there are specific steps or configurations needed to resolve this issue, please let me know. Thank you for your assistance. Best regards, Jafar |
From: Šindelka P. <sin...@tt...> - 2024-06-18 17:01:59
|
I had a look through my scenarios and I was always only sending REFER so far, never expecting it, but I don't think it should be treated in some special way. Can you post the .pcap, please? From the presence of [peer_tag_param] in your scenario I assume that it is a mid-dialog REFER, so maybe there is something wrong in the REFER message itself. Pavel Dne 18.06.2024 v 14:53 Tom Johnson napsal(a): > I am using: > > <recv request="REFER" /> > <send> > <![CDATA[ > SIP/2.0 202 Accepted > Via: SIP/2.0/TCP [remote_ip]:5060;branch=[branch] > From: <sip:91...@ps...>[peer_tag_param] > To: +14253244587 > <sip:4253244587@[local_ip]:5060;isup-oli=62>;tag=[call_number] > Call-ID: [call_id] > CSeq: 102 REFER > Content-Length: 0 > Server: INdigital-inesrp-1.26.6 > > ]]> > </send> > > However, from Wireshark, I see "603 Decline" and sipp does not get > past the waiting for REFER step. Can't seem to find any examples of > receiving a "refer". Any help would be greatly appreciated. > > > > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users -- *Ing. Pavel Šindelka * senior specialista TTC MARCONI s. r. o. Třebohostická 987/5, 100 00 Praha 10 +420 234 051 712, +420 602 355 199 sin...@tt..., www.ttc-marconi.com |
From: Tom J. <TJo...@mi...> - 2024-06-18 16:28:00
|
I am using: <recv request="REFER" /> <send> <![CDATA[ SIP/2.0 202 Accepted Via: SIP/2.0/TCP [remote_ip]:5060;branch=[branch] From: <sip:91...@ps...>[peer_tag_param] To: +14253244587 <sip:4253244587@[local_ip]:5060;isup-oli=62>;tag=[call_number] Call-ID: [call_id] CSeq: 102 REFER Content-Length: 0 Server: INdigital-inesrp-1.26.6 ]]> </send> However, from Wireshark, I see "603 Decline" and sipp does not get past the waiting for REFER step. Can't seem to find any examples of receiving a "refer". Any help would be greatly appreciated. |
From: Chandrasekaran A. <amb...@al...> - 2024-04-30 09:45:48
|
Dear all Good day ! We have the following Scenario is not working SIPP 1 user calling SIPP2 via PBX and Automated Attendant and Automated Attendant is transfer the call to Pbx and it will reach the SIPP2 extension which was registered with Pabx. SIPP1 ----> PBX---> (Automated Attendant ) -- > Transfer the call to Pabx---> SIPP - 2 Extension is not working It seems the call leg is not matching . If you have any xml file matching this scenario please send to me. Thanks Best regards, Chandrasekaran Ambalavanan |
From: GUILLAUME Ely-A. <ely...@sp...> - 2024-04-17 12:49:55
|
Hello, I'm having trouble creating a SIP trunk between my SIPp server and a Mitel MICC system. Do you have an idea of something I could be doing wrong? Ely-Anne GUILLAUME Architecte DWP Tél : +33 635 20 06 55 ely...@sp...<mailto:ely...@sp...> SPIE ICS Dir. Innovation Design et Expertise DAIDF 148 avenue Pierre Brossolette 92240 MALAKOFF www.spie-ics.com<http://www.spie-ics.com/> Ce message et toutes les pièces jointes (ci-après le "message") sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute modification, édition, utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. SPIE et ses filiales déclinent toute responsabilité au titre de ce message s'il a été altéré, déformé, falsifié, édité ou diffusé sans autorisation. This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised alteration , printing , use or dissemination is prohibited. E-mails are susceptible to alteration. SPIE nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed, falsified, printed or disseminated without authorisation. |
From: Chaigneau, N. <nic...@ca...> - 2023-11-09 15:22:49
|
Hello, I have a question related to SIPp logging. It seems that messages logged in the SIPp error file are not terminated by an end of line character. This makes it difficult to read or parse. See example below: The following events occurred: 2023-11-09 14:46:52.564911 1699541212.564911: Call-Id: (...), receive timeout on message (...) UAC:1 without label to jump to (ontimeout attribute): aborting call2023-11-09 14:46:52.565879 1699541212.565879: Dead call 1-4148690@10.118.21.108 (aborted at index 1), received 'SIP/2.0 302 Moved Temporarily^M (...) Content-Length: 0^M ^M '2023-11-09 14:46:53.564003 1699541213.564003: Call-Id:(...) In this example, the corresponding code is in call.cpp at line 2054: WARNING("Call-Id: %s, receive timeout on message %s:%d without label to jump to (ontimeout attribute): aborting call", id, curmsg->desc, curmsg->index); I don't see a \n at the end of the log message (or for any call to WARNING or ERROR functions). Shouldn't there be one ? Or should the logging function add a \n ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. |
From: Ihor O. <igo...@gm...> - 2023-11-03 13:41:46
|
Ozgur, You can try built it inside WSL on Windows. Or docker. Not really native way, but definitely something that will work Le lun. 16 oct. 2023 à 09:29, özgür pektaş <ozg...@ho...> a écrit : > Hi all, > > I've been trying to use sipp with authentication but it fails due to > openssl dependency. İ tried 3.1 and 3.2 executable installation versions > but it seems open sll was not built in it. Then i installed cygwin64 > followed instructions from here > > https://siptestingknowledge.blogspot.com/2013/12/installing-sipp-on-windows.html?m=1 > > And tried newer releases of sipp but I've never managed to get it working. > İ highly appreciate if anyone help me in this. > > Thanks > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users > -- Best regards, Ihor (Igor) |
From: özgür p. <ozg...@ho...> - 2023-10-16 07:28:17
|
Hi all, I've been trying to use sipp with authentication but it fails due to openssl dependency. İ tried 3.1 and 3.2 executable installation versions but it seems open sll was not built in it. Then i installed cygwin64 followed instructions from here https://siptestingknowledge.blogspot.com/2013/12/installing-sipp-on-windows.html?m=1 And tried newer releases of sipp but I've never managed to get it working. İ highly appreciate if anyone help me in this. Thanks |
From: Andrej M. <xm...@st...> - 2023-04-15 14:44:49
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<html><body><div>Hi.<br><br></div><div>I am trying to install IMS Bench SIPp on Linux operating system following the instructions at https://sipp.sourceforge.net/ims_bench/reference.html#Installation<br>with command:</div><div><i>svn co https://sipp.svn.sourceforge.net/svnroot/sipp/sipp/branches/ims_bench ims_bench<br></i><br></div><div>But the command failed with the message <i>"Repository moved permanently to 'https://sourceforge.net/projects/sipp/'; please relocate "</i></div><div><i><br></i>I could not find another source of the IMS Bench SIPp program.<br>Thanks for help.</div><div>Andy</div></body></html> |
From: Olivier <oza...@gm...> - 2023-02-24 10:27:50
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Hello, I've just read [1] with which I learned about -key command line parameter. 1. If I may add a comment, Debian's SIPp man page does not describe the use of this setting, very well. To improve this, would ping downstream (Debian) or upstream (SIPp) ? 2. How within a scenario file, I suppose, can I set a default value to -key parameters so that I don't need to pass a value to SIPp anytime I run SIPp ? In referenced example, how can I avoid "-key pcap_file foo.wav" ? [1] https://stackoverflow.com/questions/48952295/using-sipp-variable-inside-exec-section [2] https://manpages.debian.org/bullseye/sip-tester/sipp.1.en.html Best regards |
From: Alberto D. <alb...@mo...> - 2022-09-30 12:53:51
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Hi all, I am trying to run some scenario and I get sipp automatically (without any command to do it in the scenario file) to copy Record-Route and Route received in a Response as first headers in teh following request within the same dialog. I want to avoid that behavior but have, so far, been unable to find in the source code where sipp is doing that (I'm no good with c++ and don't really understand what is going on in the sources of sipp). In my scenario sipp is emulating a proxy behind which both users are towards some chain of servers. It's sending an INVITE, receiving it back with some Record-Route and Route, then sending a 183 to the INVITE (with a copy of the Record-Route and Route received), receiving it back and then I want to send a PRACK. The problem is that sipp automatically includes the Record-Route and Route headers received into all the messages it generates. In the responses I don't mind because the Via is going to be used for the routing; but in the Requests its a problem. I need to write into the Route the content of the Record-Route I received. I can do that with a actions and variables in the scenarios. My problem is that I need to stop sipp from automatically copying the Record-Route and Route received into the next message sent. Does anyone know where that logic is in the sources so I can comment it out? This might be something that is good for everyone else, but I would rather have to tell sipp in the scenario file exactly all headers I want in the messages sent. By the way in the recv commands I always have rrs=false in case that helps, but it doesn't help. sipp includes the Record-Route and Route received in the next send message. Any hint is welcome! Thanks! Best regards alberto |
From: Alberto D. <alb...@mo...> - 2022-09-30 12:37:14
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Hi all, I wasn't able to perform Digest-AKAv1-MD5 authentication with current sipp code from sourceforge. I tried to repair it and found something that works. I suggest here the changes if any of the people who mantain sipp want to commit them or if someone wants to use Digest-AKAv1-MD5. I also attach my auth.cpp just in case someone wants a quick and dirty hack to get it through (the file includes some ugly logging I introduced when trying to debug the situation). I guess these issues have been introduced because the way the parameters from the scenario file have changed or something similar. In the file src/auth.cpp I identify following issues: 1) line 663 where it calls Milenage f1 function f1(k,rnd,sqn,(unsigned char *) aka_AMF,xmac,op); it should have been f1(k,rnd,sqn,amf,xmac,op); because aka_AMF is actually a character string and its amf variable defined as u_char * (uint8_t?) the one that would be correct there. 2) lines 650,651,653 read like this memcpy(k,aka_K,KLEN); memcpy(amf,aka_AMF,AMFLEN); memcpy(op,aka_OP,OPLEN); But aka_K, aka_AMF, aka_OP are character strings read from the scenario Authentication field parameters and in k, amf, op we need byte arrays to pass them to the Milenage functions. I introduced some hex2uint function suggested by some stackoverflow people to resolve the issue and convert between the two things and then just called that instead of doing direct memcpy. After solving those two issues Digest-AKAv1-MD5 authentication worked. In my case because I didn't introduce many safe checks into the code it was fundamental that I had the right length in the Authentication params like even if they are 0 I had to type aka_k="00000000000000000000000000000000" and same for aka_OP and aka_AMF="0000". But then it actually works. Best regards alberto |
From: Brad Z. <bra...@gm...> - 2022-07-12 22:21:03
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Hi Johan I tried that and it didn't work. On Tue, Jul 12, 2022, 3:28 PM Johan De Clercq <jo...@de...> wrote: > Inject with csv file > > Br > > Outlook voor iOS <https://aka.ms/o0ukef> downloaden > ------------------------------ > *Van:* Brad Zerr <bra...@gm...> > *Verzonden:* Tuesday, July 12, 2022 10:03:01 PM > *Aan:* Sip...@li... <Sip...@li...> > *Onderwerp:* [Sipp-users] RTP_Streaming > > Is there a way to automatically set up an RTP_Stream scenario where it > will choose the next .wav file without having to manually change my script? > > This is the RTP_Stream portion I am using today embedded in my script and > it works great. > > <nop> > <action> > <exec rtp_stream="brad.wav,1,0" /> > </action> > </nop> > > > But if I want the next call to utilize the brad2.wav file, I have to > change my script and kick off the command again. This works fine for 1 or > 2 calls, but if I have a 1,000 calls, it will be quite tedious. I tried > building a csv file and using -inf and having each row with a new .wav > file, but SIPp didn't like that. > > > |
From: Johan De C. <jo...@de...> - 2022-07-12 21:37:17
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Inject with csv file Br Outlook voor iOS<https://aka.ms/o0ukef> downloaden ________________________________ Van: Brad Zerr <bra...@gm...> Verzonden: Tuesday, July 12, 2022 10:03:01 PM Aan: Sip...@li... <Sip...@li...> Onderwerp: [Sipp-users] RTP_Streaming Is there a way to automatically set up an RTP_Stream scenario where it will choose the next .wav file without having to manually change my script? This is the RTP_Stream portion I am using today embedded in my script and it works great. <nop> <action> <exec rtp_stream="brad.wav,1,0" /> </action> </nop> But if I want the next call to utilize the brad2.wav file, I have to change my script and kick off the command again. This works fine for 1 or 2 calls, but if I have a 1,000 calls, it will be quite tedious. I tried building a csv file and using -inf and having each row with a new .wav file, but SIPp didn't like that. |
From: Brad Z. <bra...@gm...> - 2022-07-12 20:03:19
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Is there a way to automatically set up an RTP_Stream scenario where it will choose the next .wav file without having to manually change my script? This is the RTP_Stream portion I am using today embedded in my script and it works great. <nop> <action> <exec rtp_stream="brad.wav,1,0" /> </action> </nop> But if I want the next call to utilize the brad2.wav file, I have to change my script and kick off the command again. This works fine for 1 or 2 calls, but if I have a 1,000 calls, it will be quite tedious. I tried building a csv file and using -inf and having each row with a new .wav file, but SIPp didn't like that. |
From: Šindelka P. <sin...@tt...> - 2022-05-20 17:35:59
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Hello, on Windows, try "HxD Hex Editor" or "Hexedit" (I use the latter one, but that doesn't mean it is better, I just came across it first years ago when I was looking for such a tool). For Linux, use xxd to hexdump a file, edit it using any text editor, and use xxd -r to convert it back to binary. But all of these are dangerous weapons, so use them with a lot of care. To adjust the RTP headers properly, you need to open the pcap file first, make yourself familiar with the various fields of the RTP headers, understand the relationship between them, and also understand the .pcap format structure. The ability of Wireshark to highlight a field in the hex dump pane when you click it in the dissection pane should help you find what to change using the hex editor. But capturing DTMF generated by a phone and using Wireshark to filter out the necessary packets may still be a faster way to your goal. In dtmf_2833_1.pcap, there are 10 packets; in all of them, you have to change the "sequence number" field to grow monotonously by 1 and start from a higher value than the last one in the file dtmf_2833_2.pcap; the "timestamp" value must be the same in all of them (for telephone-event packets in particular - this is not the case for normal audio or video codecs), but it has to be higher than the timestamp value in dtmf_2833_2.pcap. So rather than editing the pcap file manually, it may be worthier to write a script that will generate .pcap files as needed for the particular test scenarios. P. Dne 20.05.2022 v 12:20 Vinayak Makwana napsal(a): > Hi > Thanks for your response > My scenario it's as below: > first I am sending DTMF 1 then DTMF 2 and then DTMF 1 again . but 3rd > DTMF 1 is not detected in Freeswitch. > Can you please give me suggestions like how can i change sequence > number or RTP timestamp? and > Can you provide me with a binary editor link from where I can make > changes in the PCAP file? > > Regards > Vinayak Makwana > > On Wed, May 18, 2022 at 9:22 PM Šindelka Pavel <sin...@tt...> wrote: > > I would assume that the pcap player invoked by SIPp does not > adjust the > RTP sequence numbers and/or RTP timestamps when playing the .pcap > files, > so if the receiving RTP stack checks them, it considers all the > packets > of the second instance to arrive late and ignores them. But it is > weird > if a DTMF received in one call affects its processing in another > call, > is that indeed the case? > > If my assumption is correct, you'll have to create a copy of the pcap > file and edit the sequence numbers and/or timestamps using some > editor > of binary files, or capture some real DTMF, or maybe modify the > source > code of the pcap player to take care of this. > > Pavel > > > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users > > > *https://www.ecosmob.com/itexpo/ <https://www.ecosmob.com/itexpo/> > * > * > * > *Disclaimer** > * > In addition to generic Disclaimer which you have agreed on our > website, any views or opinions presented in this email are solely > those of the originator and do not necessarily represent those of the > Company or its sister concerns. Any liability (in negligence, contract > or otherwise) arising from any third party taking any action, or > refraining from taking any action on the basis of any of the > information contained in this email is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for > the use of the addressee(s) and contains information that is > PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, > distribution, or copying of this communication is prohibited. Please > inform originator if you have received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of > viruses, trojans, similar or new contaminants/malware, interceptions > or interference, and may not be compatible with your systems. You > shall carry out virus/malware scanning on your own before opening any > attachment to this e-mail. The sender of this e-mail and Company > including its sister concerns shall not be liable for any damage that > may incur to you as a result of viruses, incompleteness of this > message, a delay in receipt of this message or any other computer > problems. |
From: Vinayak M. <vin...@ec...> - 2022-05-20 11:23:42
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Hi Thanks for your response My scenario it's as below: first I am sending DTMF 1 then DTMF 2 and then DTMF 1 again . but 3rd DTMF 1 is not detected in Freeswitch. Can you please give me suggestions like how can i change sequence number or RTP timestamp? and Can you provide me with a binary editor link from where I can make changes in the PCAP file? Regards Vinayak Makwana On Wed, May 18, 2022 at 9:22 PM Šindelka Pavel <sin...@tt...> wrote: > I would assume that the pcap player invoked by SIPp does not adjust the > RTP sequence numbers and/or RTP timestamps when playing the .pcap files, > so if the receiving RTP stack checks them, it considers all the packets > of the second instance to arrive late and ignores them. But it is weird > if a DTMF received in one call affects its processing in another call, > is that indeed the case? > > If my assumption is correct, you'll have to create a copy of the pcap > file and edit the sequence numbers and/or timestamps using some editor > of binary files, or capture some real DTMF, or maybe modify the source > code of the pcap player to take care of this. > > Pavel > > > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users > -- * <https://www.ecosmob.com/itexpo/> * * * *Disclaimer** * In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. |
From: Šindelka P. <sin...@tt...> - 2022-05-18 15:51:26
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I would assume that the pcap player invoked by SIPp does not adjust the RTP sequence numbers and/or RTP timestamps when playing the .pcap files, so if the receiving RTP stack checks them, it considers all the packets of the second instance to arrive late and ignores them. But it is weird if a DTMF received in one call affects its processing in another call, is that indeed the case? If my assumption is correct, you'll have to create a copy of the pcap file and edit the sequence numbers and/or timestamps using some editor of binary files, or capture some real DTMF, or maybe modify the source code of the pcap player to take care of this. Pavel |
From: Vinayak M. <vin...@ec...> - 2022-05-18 13:55:25
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Hello All, I am trying to send multiple DTMF to the FS server.but sometimes I need to send the same DTMF as per requirement . Here's my scenario First I send DTMF as 1 and again I need to send the same DTMF in between calls.in Freeswitch side I got the first DTMF but the second time I got NULL. So can anyone help me with this issue? I am using 3.6.1 Sipp version Here's My sipp script: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "+9892545555.dtd"> <scenario name="UAC with media"> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: <sip:1013@ [local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number] To: <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: <sip:1013@[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] 1.2.3.4 t=0 0 m=audio 8000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="180" optional="true"> </recv> <recv response="183" optional="true"> </recv> <recv response="200" rrs="true" rtd="true" crlf="true"> </recv> <pause milliseconds="500"/> <send> <![CDATA[ ACK [next_url] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] [routes] From: <sip:1013@ [local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number] To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: <sip:1013@[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause milliseconds="8000"/> <!-- Play an out of band DTMF '1' --> <nop> <action> <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/> </action> </nop> <pause milliseconds="12000"/> <nop> <action> <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/> </action> </nop> <pause milliseconds="8000"/> <!-- The 'crlf' option inserts a blank line in the statistics report. --> <send retrans="500"> <![CDATA[ BYE [next_url] SIP/2.0 [routes] Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: <sip:1013@ [local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number] To: <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: <sip:1013@[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true"> </recv> <!-- definition of the response time repartition table (unit is ms) --> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> <!-- definition of the call length repartition table (unit is ms) --> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> Regards Vinayak Makwana -- * <https://www.ecosmob.com/itexpo/> * * * *Disclaimer** * In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. |
From: Šindelka P. <sin...@tt...> - 2022-04-20 13:38:31
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Look into the manual how to handle the authentication dynamically. The contents of the Authorization header cannot be static as it is a response to a dynamically changing challenge used to prove that the UAC knows the shared secret. So the correct procedure is that you send a request with no Authorization header at all (or one with an old contents), and if the UAS sends you a 401 with an authentication challenge, you re-send the same request with an Authorization header with the correct answer to the challenge. The UAC scenario must account for this possibility, it is not done automatically. I.e. one possible implementation: send INVITE without Authorization recv 100, 180, 183, ... optional recv 401 optional next auth-invite recv 200 next call-success label auth-invite send INVITE with Authorisation recv 100, 180, 183, ... optional recv 401 optional next authentication-failure recv 200 label call-success ... label authentication-failure Pavel |