[Sipp-users] Different port for remote side
Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060) Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] (no subject)
Hi, I am using SIPP to generate a call with RTP media. But the problem is that packets are not reaching the network. I am using Wireshark to see the captured packets on the network. I have installed both libnet and libpcap libraries which are essential for the handling of RTP packets. I have compiled SIPp to enable it with PCAP play using the following: make pcapplay But still the issue is not resolved. It would be a great help if somebody can tell me the solution for this and also find attached the scenario file. Regards, Shalu Dhamija uac_pcap_29636_screen.log Description: Binary data - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Issue: RTP packets not reaching the network
_ Hi, I am using SIPP to generate a call with RTP media. But the problem is that packets are not reaching the network. I am using Wireshark to see the captured packets on the network. I have installed both libnet and libpcap libraries which are essential for the handling of RTP packets. I have compiled SIPp to enable it with PCAP play using the following: make pcapplay But still the issue is not resolved. It would be a great help if somebody can tell me the solution for this and also find attached the scenario file. Regards, Shalu Dhamija uac_pcap_29636_screen.log Description: Binary data - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Different port for remote side
SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060) Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Issue faced with updating filed value of csv injection file
No. SIPp's internal timing loop will go haywire trying to catch up when you resume it. Charles Madiha Shahid [EMAIL PROTECTED] 09/24/2008 01:05 AM To Peter Higginson [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] Issue faced with updating filed value of csv injection file Thanks Peter for the suggestion. Thats right, the scenario I'm using works for one call only. Would it be a good idea to pause the SIPP process using 'kill -SIGSTOP' command on linux and resume it after the media transfer gets completed by using the 'kill -SIGCONT' command? Regards, Madiha On Wed, Sep 24, 2008 at 1:36 AM, Peter Higginson [EMAIL PROTECTED] wrote: Madiha, The mechanism you have described looks like it only works with one call. If that is the case you could exit SIPP (saving any context and the Call-ID of course) and re-enter it to continue the call after the media is done. The alternative we did at Newport Networks was to start and stop the external media generator from the SIPP process. That method will (and did) work for multiple simultaneous calls and you then use something like a pause to control the length of the media generation. Peter Higginson Date: Tue, 23 Sep 2008 21:35:33 +0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] Issue faced with updating filed value of csv injection file Hi, Thanks for the reply Charles. Does anyone know a workaround to this problem. Is there a way to induce a variable pause at the server side SIPp such that the the file execution of the server side resumes only after the media transfer gets completed. Regards, Madiha On Tue, Sep 23, 2008 at 4:49 PM, Charles P Wright [EMAIL PROTECTED] wrote: You can not update the value of CSV fields after starting SIPp. Charles Madiha Shahid [EMAIL PROTECTED] 09/23/2008 03:02 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Issue faced with updating filed value of csv injection file Hi all, Description: I am writing a scenario in SIPp that allows media transfer between calls using an external utility (Gstreamer). The external utility gets called by running it through exec command.I want to induce a pause at the sender side so that media transfer gets completed before further messages can be tranfered between SIPp client and server. This is how Im trying to do it. I use the -inf switch and provide a csv file as input to the server side sipp command The [field0] in this csv file has vale 1. When file transfer gets completed, value '1' written in this file is replaced with value '10' as written by an external application. The SIPp server, keeps monitoring the [field0] value to check if the the file has been updated so that it can proceed further. However, even though the value in the csv file is replaced, it is not updated in the [field0]. [field0] still has the old value which keeps the scenario in a loop for ever. Please let me know if this is expected? Is there a workaround to this problem? Thanks, Madiha Here is the part of the code at the server side that produces this issue: ** ** nop action exec command=./gst-sender.sh/ /action /nop label id=8/ nop action log message=entered label 8/ /action /nop pause milliseconds=1/ nop action !-- Assign the value in field0 of the CSV file to a $3. -- assignstr assign_to=3 value=[field0] / log message=Value written in file is [$3]/ todouble assign_to=4 variable=3 / log message=Value written in file converted is [$4]/ test assign_to=5 variable=4 compare=not_equal value=10 / log message=Result of compare is [$5]/ /action /nop nop next=8 test=5/ nop action log message=exiting label 8/ - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users Get Hotmail on your mobile from Vodafone Try it Now! - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing
Re: [Sipp-users] Different port for remote side
Hi Romain, I experienced a similar issue but it was due to 1 white spaces between CSEQ number and CSEQ method. Can you check the 180/100 to see how many spaces exist there? Ex. CSeq: 1 INVITE Thanks! Romain Gautier wrote: Thank you for your reply. Indeed SIPp logs successfullly the "TRYING" within the embedded uac scenario. Nevertheless, using my own scenario, SIPp logs the TRYING in the errors log file: the TRYING cannot be mapped to a known SIPp call, although the Call-ID is correct. Should it be an encoding issue of my scenario file? Cdt Romain 2008/9/24 Charles P Wright [EMAIL PROTECTED] SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles "Romain Gautier" rhum1.gt@gmail.com wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060)Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=""> ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=""> ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] -aa option does not send out 200OK(NOTIFY)
Hi All, I am using SIPp3.1 to Register a UE with SUBSCRIBE/NOTIFY. I want SIPp to generate an automatic 200OK response to the NOTIFY message. I tried using -aa option, however, when I look at the trace - SIPp does not send out a 200 OK of NOTIFY. Do I need to add anything besides the -aa on the command line? thx, Kalpesh. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Different port for remote side
Romain, CoolSounds like you figured it out and it had nothing to do with the white space. Romain Gautier wrote: Hi Scott, I found 1 white space between CSEQ number and CSEQ method in each message. Rgds 2008/9/24 Scott Page [EMAIL PROTECTED] Hi Romain, I experienced a similar issue but it was due to 1 white spaces between CSEQ number and CSEQ method. Can you check the 180/100 to see how many spaces exist there? Ex. CSeq: 1 INVITE Thanks! Romain Gautier wrote: Thank you for your reply. Indeed SIPp logs successfullly the "TRYING" within the embedded uac scenario. Nevertheless, using my own scenario, SIPp logs the TRYING in the errors log file: the TRYING cannot be mapped to a known SIPp call, although the Call-ID is correct. Should it be an encoding issue of my scenario file? Cdt Romain 2008/9/24 Charles P Wright [EMAIL PROTECTED] SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles "Romain Gautier" rhum1.gt@gmail.com wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060)Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=""> ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=""> ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Different port for remote side
You should post your message trace and error trace to the list. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 10:13:16 AM: Thank you for your reply. Indeed SIPp logs successfullly the TRYING within the embedded uac scenario. Nevertheless, using my own scenario, SIPp logs the TRYING in the errors log file: the TRYING cannot be mapped to a known SIPp call, although the Call-ID is correct. Should it be an encoding issue of my scenario file? Cdt Romain 2008/9/24 Charles P Wright [EMAIL PROTECTED] SIPp should be able to receive and process the message from a different port. I would use a combination of packet capture and -trace_msg to make sure the packet arrives at SIPp. Charles Romain Gautier [EMAIL PROTECTED] wrote on 09/24/2008 04:37:54 AM: Hi, I am trying to use SIPp as an UAC, using a scenario. My problem is that SIPp does not seem to recognize the SIP responses. Indeed the 180 trying is not taken into account by SIPp. The remote part use a different port in order to send its response, is it a problem? Is there any turn-around? SIPp(5060) Remote INVITE (port 5060) port 5060 port 5060 TRYING from port (dynamic port) Thank you for your answer. Cordially - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Issue faced with updating filed value of csv injection file
Yes, I got the following errors when I tried to pause the SIPP process using 'kill -SIGSTOP' command on linux and resumed it after the media transfer got completed by using the 'kill -SIGCONT' command. The minor watchdog timer 500ms has been tripped (1200), 120 trips remaining.. Resetting watchdog timer trigger counts, as it has not been triggered in over 600041ms.. Apparently the flow of messages and media was successful. I found no help in the SIPp documentation regarding the working of this timer so I am not sure how this error may affect various scenarios using the pause approach I have mentioned. Can you please suggest any relevant reference documentation I could use. Can you kindly recommend a better solution to this problem. Regards, Madiha On Wed, Sep 24, 2008 at 6:55 PM, Charles P Wright [EMAIL PROTECTED]wrote: No. SIPp's internal timing loop will go haywire trying to catch up when you resume it. Charles Madiha Shahid [EMAIL PROTECTED] 09/24/2008 01:05 AM To Peter Higginson [EMAIL PROTECTED] cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] Issue faced with updating filed value of csv injection file Thanks Peter for the suggestion. Thats right, the scenario I'm using works for one call only. Would it be a good idea to pause the SIPP process using 'kill -SIGSTOP' command on linux and resume it after the media transfer gets completed by using the 'kill -SIGCONT' command? Regards, Madiha On Wed, Sep 24, 2008 at 1:36 AM, Peter Higginson [EMAIL PROTECTED] wrote: Madiha, The mechanism you have described looks like it only works with one call. If that is the case you could exit SIPP (saving any context and the Call-ID of course) and re-enter it to continue the call after the media is done. The alternative we did at Newport Networks was to start and stop the external media generator from the SIPP process. That method will (and did) work for multiple simultaneous calls and you then use something like a pause to control the length of the media generation. Peter Higginson Date: Tue, 23 Sep 2008 21:35:33 +0500 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] CC: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] Issue faced with updating filed value of csv injection file Hi, Thanks for the reply Charles. Does anyone know a workaround to this problem. Is there a way to induce a variable pause at the server side SIPp such that the the file execution of the server side resumes only after the media transfer gets completed. Regards, Madiha On Tue, Sep 23, 2008 at 4:49 PM, Charles P Wright [EMAIL PROTECTED] wrote: You can not update the value of CSV fields after starting SIPp. Charles Madiha Shahid [EMAIL PROTECTED] 09/23/2008 03:02 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] Issue faced with updating filed value of csv injection file Hi all, Description: I am writing a scenario in SIPp that allows media transfer between calls using an external utility (Gstreamer). The external utility gets called by running it through exec command.I want to induce a pause at the sender side so that media transfer gets completed before further messages can be tranfered between SIPp client and server. This is how Im trying to do it. I use the -inf switch and provide a csv file as input to the server side sipp command The [field0] in this csv file has vale 1. When file transfer gets completed, value '1' written in this file is replaced with value '10' as written by an external application. The SIPp server, keeps monitoring the [field0] value to check if the the file has been updated so that it can proceed further. However, even though the value in the csv file is replaced, it is not updated in the [field0]. [field0] still has the old value which keeps the scenario in a loop for ever. Please let me know if this is expected? Is there a workaround to this problem? Thanks, Madiha Here is the part of the code at the server side that produces this issue: ** ** nop action exec command=./gst-sender.sh/ /action /nop label id=8/ nop action log message=entered label 8/ /action /nop pause milliseconds=1/ nop action !-- Assign the value in field0 of the CSV file to a $3. -- assignstr assign_to=3 value=[field0] / log message=Value written in file is [$3]/ todouble assign_to=4 variable=3 / log message=Value written in file converted is [$4]/ test assign_to=5 variable=4 compare=not_equal value=10 / log message=Result of compare is [$5]/ /action /nop nop next=8 test=5/ nop action log
[Sipp-users] Issue with the register request.
Hi I am trying to register a set of users to my test PBX using SIPP. However I getting some issues as below 1. On the captures I get UDP checksum incorrect for all the SIP messages that I send out 2. Once I send out the register request I get a 401 unauthorized to which I send a register with Authentication information however I get a 401 again. 3. Also I noticed on the captures that the second register request does not have a content length information. Also the capturing tool(wireshark) indicates malformed SIP packet. I am attaching my xml csv file and two capture files, the first capture file has sucessful registration using a normal phone. the second capture has a SIPP registration failure captures. Thanking you all in advance for your help -Harsh ?xml version=1.0 encoding=ISO-8859-1 ? !DOCTYPE scenario SYSTEM sipp.dtd scenario name=UAC Basic Registration label id=0 / send retrans=500 start_rtd=true ![CDATA[ REGISTER sip:[field1] SIP/2.0 From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field0] sip:[EMAIL PROTECTED] Call-ID: [call_id] CSeq: [cseq] REGISTER Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Supported: 100rel,replaces User-Agent: Vertical SIPP TEST TOOL Contact: [field0] sip:[EMAIL PROTECTED]:[local_port];expires=3600 Content-Length: 0 ]] /send recv response=200 rtd=true next=3 optional=true / recv response=403 rtd=true next=2 optional=true / recv response=404 rtd=true next=2 optional=true / recv response=401 rtd=true next=1 auth=true / label id=1 / send retrans=500 start_rtd=true ![CDATA[ REGISTER sip:[field1] SIP/2.0 From: [field0] sip:[EMAIL PROTECTED];tag=[call_number] To: [field0] sip:[EMAIL PROTECTED] Call-ID: [call_id] CSeq: [cseq] REGISTER Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: [field0] sip:[EMAIL PROTECTED]:[local_port];expires=3600 [field2] Content-Length: 0 ]] /send recv response=400 rtd=true next=2 optional=true / recv response=403 rtd=true next=2 optional=true / recv response=404 rtd=true next=2 optional=true / recv response=200 rtd=true next=3 crlf=true / label id=2 / nop action exec int_cmd=stop_call / /action /nop label id=3 / !-- Definition of the response time repartition table (unit is ms). -- ResponseTimeRepartition value=30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500 / !-- Definition of the call length repartition table (unit is ms). -- CallLengthRepartition value=30, 50, 80, 100, 150, 200, 300, 400, 500, 600, 700, 1000, 1200, 1400, 1500 / /scenario SEQUENTIAL 103;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=103]; 104;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=104]; 105;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=105]; 106;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=106]; 107;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=107]; 108;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=108]; 109;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=109]; 110;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=110]; 111;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=111]; 112;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=112]; 113;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=113]; 114;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=114]; 115;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=115]; 116;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=116]; 117;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=117]; 118;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=118]; 119;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=119]; 120;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=120]; 121;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=121]; 122;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=122]; 123;192.168.1.1:5060;[authentication [EMAIL PROTECTED] password=123]; register_simple_call Description: Binary data sipp_register Description: Binary data - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] SIP Clients supporting EVRC Codec
Hi, Do anyone know any SIP soft clients or hard phones (both freeware or for purchase) which supports the variants of EVRC codecs? Pls let me know the details of the same Thanks, - Sairam S - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
