sr-users September 2003

sr-users@lists.kamailio.org

86 participants
209 discussions

[Serusers] Accounting and parallel/sequential forking

by Guilherme Dal Pizzol

ello, I'm trying to generate call accounting in parallel and sequential forking, but I think I'm missing something. My ser.cfg is attached. When I perform a call (parallel fork / sequential fork) I'm getting only one miss report, for example: 1 - SER receives a INVITE 2 - INVITE is sent to A and B (parallel) 3 - A fails with 503 4 - B fails with 408 (TM generates 408) 5 - SER logs only 408!!! How about 503? Is there any way to account both misses? 6 - As both calls failed, SER will continue with failure_route[1] (sequential) that also fails with 408!! 7 - No account is generated! 8 - As the last call failed (failure_route[1]), it will continue with failure_route[2] (sequential), that also fails with 408!! 9 - At this point, SER will account the last failure only if failed_transactions is set to "1". Is this the correct behaviour? Thanks in advance. Guilherme.

20 years, 8 months

[Serusers] :: nathelper vs jasomi

by abdoul

Hello, nathelper module is it something similar to what Jasomi Network is doing ? (PeerPoint Centrex) If not is it possible for me to start from what you've done to obtain something similar than Jasomi? Thanks in advance for your answer Regards Abdoul

20 years, 8 months

RE: [Serusers] NAT & RTPPRoxy

by Gary Brewer

>BTW, the webpage says that you can find out the IP and ports used by >yourself if the NAT doesn't support UPnP and make the API to use it. Yes I believe you can create an IRTCPortManager and use GetMapping to specify your AV ports and SIP Signalling) when you are starting a session (http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rt c/irtcportmanager_getmapping.asp) However the registration process (not the INVITE) does However the registration process does not appear to allow you to communicate your external address, I don't see where it allows you to do that anywhere in the API. The problem is I don't want to support uPnP at all, I want to support non-uPnP NATS. Thanks for the FCP link, I will look into that! Gary -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: 03 September 2003 11:39 To: Gary Brewer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] NAT & RTPPRoxy BTW, the webpage says that you can find out the IP and ports used by yourself if the NAT doesn't support UPnP and make the API to use it. BTW2: I am aware of one IP phone that supports UPnP: http://www.act-tel.com.tw Those guys had UPnP testbed at the last SIPIt. Jan. On 03-09 11:14, Gary Brewer wrote: > Thanks Jan, > > I should have made myself a little bit clearer; I am using the RTC 1.2 API > (http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rt > c/real_time_communications_rtc_client_start_page.asp) > > What I find the most interesting is - > http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rtc > /traversal_of_upnp_enabled_nats.asp specifically the last sentence of the > third paragraph. Why I can't provide it my external IP address and port on > the NAT is annoying! I wonder if it would be possible to modify the REGISTER > message on the way to the SIP server and change the IP:Port to my external > NAT address mapping - or get NATHelper to do this for me. > > Gary > > > > > > > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: 03 September 2003 10:34 > To: Gary Brewer > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] NAT & RTPPRoxy > > Hello, comments inline. All the NAT stuff is quite complicated and > whether it would work or not depends on many factors. > > On 03-09 09:55, Gary Brewer wrote: > > Hi, > > > > I have come across a similar problem. I want to use the A/V facilities of > > Windows Messenger if one or both of my clients are behind a NAT. I realise > > that it seems to be impossible to get this to work if the NAT is > symmetric. > > (See: RFC3489 "Applicability Statement") > > That depends. It could work even with symmetric NAT if the client you > are using does support symmetric signalling and RTP (I am not sure Windows > > Messenger is). That means the user agent must be ready to receive SIP > requests and responses on the same port which was used as the source port > for sending SIP messages. Also it must support symmetric RTP to make > media work. > > Also the user agent must create REGISTERs containing public IP of the > NAT (can be determined using STUN, for example), or you would have to > use nathelper module on the server. > > > If only one of my clients is behind a NAT then it would seem I would have > to > > communicate my NATs external address and port mapping to the non-NAT'd > > client (possibly with the help of STUN) in my SIP Invite SDP message. I > > Yes, but Windows Messenger doesn't support STUN. > > > would also have to setup UDP mappings for SIP, RTP/RTCP Audio Video on my > > NAT. Are my A/V port mappings also included in the Invite SDP message? > > In case of symmetrict RTP, the client in the public internet will > ignore what it receives in SDP and will send media back to the same IP > and port from which it comes from the other side. That makes the > communication through a NAT possible, but the client behind the NAT > must send first media package (that packet will open a pinhole in the > NAT). Also both sides must support symmetrict RTP and the client > behind the NAT must signal that it is using this approach. > > If the clients do not support symmetric signalling, then some kind of > "NAT configuration" would be necesarry. > > For example, many user agents can be configured to use ports from a > specified port range only for RTP (let's say ports 10000-10100). You > can then configure your NAT box to forward all the ports from the > range back to your user agent. > > > If both clients are NAT'd then what is the approach? I don't see how I > > register with the SIP server using an external NAT address (my guess is > this > > is what I would have to use if I wanted anyone on the other side of the > NAT > > to be able to see me). MSFT have seemed to got around this problem by > > recommended everyone to use uPnP enabled NATs, which will automatically > bind > > to an external address on the NAT and, I assume, use this when they > register > > with the SIP server. > > If both clients are behind NATs then you would probably have to use an > RTP proxy which will be placed in the public internet. > > > RTPProxy is here https://demo.portaone.com/~sobomax/PortaSIP/ how does > > RTPProxy help in the NAT situation, does it at all? > > See the previous insertion. It could help in the case when both > clients are behind NATs and it could possibly help in some other cases > too. > > > Please correct me where I am wrong, I am still trying to get my head > around > > all of this! Has anyone successfully been able to get any of the scenarios > > above working (with AV)? > > There is a couple of people who are running their clients behind NATs > on the mailing list, so I hope they could give you some advice. > > Jan. > > > > > This email may contain material that is confidential and/or privileged. It > is for the sole use of the intended recipient. Any review, reliance or > distribution by others of the e-mail or its contents or forwarding without > express permission is strictly prohibited. No contractual undertakings are > accepted by virtue of transmission of this e-mail unless expressly stated > otherwise in the text of the e-mail itself. If you are not the intended > recipient, please contact the sender and delete all copies. Contact us > directly or via our website at www.rawcommunications.com. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers This email may contain material that is confidential and/or privileged. It is for the sole use of the intended recipient. Any review, reliance or distribution by others of the e-mail or its contents or forwarding without express permission is strictly prohibited. No contractual undertakings are accepted by virtue of transmission of this e-mail unless expressly stated otherwise in the text of the e-mail itself. If you are not the intended recipient, please contact the sender and delete all copies. Contact us directly or via our website at www.rawcommunications.com.

20 years, 8 months

RE: [Serusers] NAT & RTPPRoxy

by Gary Brewer

Thanks Jan, I should have made myself a little bit clearer; I am using the RTC 1.2 API (http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rt c/real_time_communications_rtc_client_start_page.asp) What I find the most interesting is - http://msdn.microsoft.com/library/default.asp?url=/library/en-us/rtcclnt/rtc /traversal_of_upnp_enabled_nats.asp specifically the last sentence of the third paragraph. Why I can't provide it my external IP address and port on the NAT is annoying! I wonder if it would be possible to modify the REGISTER message on the way to the SIP server and change the IP:Port to my external NAT address mapping - or get NATHelper to do this for me. Gary -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: 03 September 2003 10:34 To: Gary Brewer Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] NAT & RTPPRoxy Hello, comments inline. All the NAT stuff is quite complicated and whether it would work or not depends on many factors. On 03-09 09:55, Gary Brewer wrote: > Hi, > > I have come across a similar problem. I want to use the A/V facilities of > Windows Messenger if one or both of my clients are behind a NAT. I realise > that it seems to be impossible to get this to work if the NAT is symmetric. > (See: RFC3489 "Applicability Statement") That depends. It could work even with symmetric NAT if the client you are using does support symmetric signalling and RTP (I am not sure Windows Messenger is). That means the user agent must be ready to receive SIP requests and responses on the same port which was used as the source port for sending SIP messages. Also it must support symmetric RTP to make media work. Also the user agent must create REGISTERs containing public IP of the NAT (can be determined using STUN, for example), or you would have to use nathelper module on the server. > If only one of my clients is behind a NAT then it would seem I would have to > communicate my NATs external address and port mapping to the non-NAT'd > client (possibly with the help of STUN) in my SIP Invite SDP message. I Yes, but Windows Messenger doesn't support STUN. > would also have to setup UDP mappings for SIP, RTP/RTCP Audio Video on my > NAT. Are my A/V port mappings also included in the Invite SDP message? In case of symmetrict RTP, the client in the public internet will ignore what it receives in SDP and will send media back to the same IP and port from which it comes from the other side. That makes the communication through a NAT possible, but the client behind the NAT must send first media package (that packet will open a pinhole in the NAT). Also both sides must support symmetrict RTP and the client behind the NAT must signal that it is using this approach. If the clients do not support symmetric signalling, then some kind of "NAT configuration" would be necesarry. For example, many user agents can be configured to use ports from a specified port range only for RTP (let's say ports 10000-10100). You can then configure your NAT box to forward all the ports from the range back to your user agent. > If both clients are NAT'd then what is the approach? I don't see how I > register with the SIP server using an external NAT address (my guess is this > is what I would have to use if I wanted anyone on the other side of the NAT > to be able to see me). MSFT have seemed to got around this problem by > recommended everyone to use uPnP enabled NATs, which will automatically bind > to an external address on the NAT and, I assume, use this when they register > with the SIP server. If both clients are behind NATs then you would probably have to use an RTP proxy which will be placed in the public internet. > RTPProxy is here https://demo.portaone.com/~sobomax/PortaSIP/ how does > RTPProxy help in the NAT situation, does it at all? See the previous insertion. It could help in the case when both clients are behind NATs and it could possibly help in some other cases too. > Please correct me where I am wrong, I am still trying to get my head around > all of this! Has anyone successfully been able to get any of the scenarios > above working (with AV)? There is a couple of people who are running their clients behind NATs on the mailing list, so I hope they could give you some advice. Jan. This email may contain material that is confidential and/or privileged. It is for the sole use of the intended recipient. Any review, reliance or distribution by others of the e-mail or its contents or forwarding without express permission is strictly prohibited. No contractual undertakings are accepted by virtue of transmission of this e-mail unless expressly stated otherwise in the text of the e-mail itself. If you are not the intended recipient, please contact the sender and delete all copies. Contact us directly or via our website at www.rawcommunications.com.

20 years, 8 months

[Serusers] NAT & RTPPRoxy

by Gary Brewer

Hi, I have come across a similar problem. I want to use the A/V facilities of Windows Messenger if one or both of my clients are behind a NAT. I realise that it seems to be impossible to get this to work if the NAT is symmetric. (See: RFC3489 "Applicability Statement") If only one of my clients is behind a NAT then it would seem I would have to communicate my NATs external address and port mapping to the non-NAT'd client (possibly with the help of STUN) in my SIP Invite SDP message. I would also have to setup UDP mappings for SIP, RTP/RTCP Audio Video on my NAT. Are my A/V port mappings also included in the Invite SDP message? If both clients are NAT'd then what is the approach? I don't see how I register with the SIP server using an external NAT address (my guess is this is what I would have to use if I wanted anyone on the other side of the NAT to be able to see me). MSFT have seemed to got around this problem by recommended everyone to use uPnP enabled NATs, which will automatically bind to an external address on the NAT and, I assume, use this when they register with the SIP server. RTPProxy is here https://demo.portaone.com/~sobomax/PortaSIP/ how does RTPProxy help in the NAT situation, does it at all? Please correct me where I am wrong, I am still trying to get my head around all of this! Has anyone successfully been able to get any of the scenarios above working (with AV)? Regards, Gary Brewer -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: 03 September 2003 00:35 To: Dan Fernandez Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] No audio Hello, is one (or both) phones behind a NAT ? If so you will need to configure your nat to let the media traffic through. If not, then we will need more information. Jan. On 02-09 18:55, Dan Fernandez wrote: > Greetings from So. America! > > I just begun playing with SER. So far I can place calls within a LAN. I > have SER running on a box, with one public and one private IP. My problem > right now is that if I try to call from a UA connected via a dialup, there > is no audio in either end. The setup of the call seems fine (the phones ring > OK). I have disable the firewall and still no luck. Do I need to install > rtpproxy? Where can I download it?(the links I´ve seen don´t work) > > Any help would be greatly appreciated. > > Rgds > Dan > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers This email may contain material that is confidential and/or privileged. It is for the sole use of the intended recipient. Any review, reliance or distribution by others of the e-mail or its contents or forwarding without express permission is strictly prohibited. No contractual undertakings are accepted by virtue of transmission of this e-mail unless expressly stated otherwise in the text of the e-mail itself. If you are not the intended recipient, please contact the sender and delete all copies. Contact us directly or via our website at www.rawcommunications.com.

20 years, 8 months

[Serusers] register problem on kphone, ser, and Debian

by Bob Horvath

I'm trying to get ser working, and have been using kphone as my client. It registers numerous times, but doesn't seem to see the 200 OK. Has anyone experienced this before? I am sure it is a stupid newbie problem. I can attach the kphone log if that helps. I wasn't sure how this list is with attachments.

20 years, 8 months

[Serusers] Newbie question about uri rewriting

by Marc Archer

Hi, I couldn't find any info on how to search the mailing list archives for the answer to my question, so I'll put it here. Hopefully someone can help me out. Does ser support URI rewriting? In my scenario we have a pstn gateway that will formulate a sip invite request uri with the phone number as the host portion e.g. sip:5551231234@192.168.0.2 where 192.168.0.2 is the address of the ser proxy What I need to be able to do is convert that 5551231234(a)192.168.0.2 to host@domain presumably using ENUM/DNS type services. Does ser support this? Does anyone know a free DNS server that supports ENUM? Thanks for the help. Cheers, Marc Marc Archer Aastra Technologies Ltd, 8 Federal Street, Billerica, MA 01821 Tel: +1 978-436-4251 Fax: +1 978-436-4233 Web: www.aastra.com

20 years, 8 months

[Serusers] installation issue with mysql.so

by Gavin Bensom

Hello all, Today I tried installing mysql, php, and ser 0.8.11 [released today] from source. After spending a lot of time, I still can't get past the error shown in the messages file attached. The response at the command line is "Bad config file, 1 error" The mysql.so module does indeed exist in the directory specified in the ser.cfg file. I've checked it over and over again. Mysql is running. Ser database tables exist. I don't know what else to do. Thanks, G. --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo. Sep 1 21:43:01 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:43:01 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:46:26 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:46:26 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:46:28 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:46:28 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:46:29 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:46:29 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:46:29 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:46:29 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:46:30 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:46:30 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 21:49:52 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 21:49:52 jiffypop ser: parse error (29,13-49): failed to load module Sep 1 22:04:17 jiffypop ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory Sep 1 22:04:17 jiffypop ser: parse error (29,13-49): failed to load module

20 years, 8 months

[Serusers] Grandstream Phones

by Gavin Bensom

Hello, Has anyone used the Grandstream BudgeTone-100 phones with SER? When I try to transfer a call the call gets disconnected. The grandstream documentation shows the transfer protocol here: http://www.grandstream.com/user_manuals/budgetone100.pdf [pg 15,16 of the .pdf] Any ideas what is happening? Is the Grandstream protocol compliant with what SER expects for transferring calls? Also, does SER support the 484 Incomplete address response? The BudgeTone phone can be set to send an INVITE after every button press, and will continue to do so if the server response with incomplete address instead of 404 not found. Finally, if users passwords are set to (null) will a phone that registers with a username but no password successfully register? [I haven't tried it yet]. Thanks, G. --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.

20 years, 8 months

[Serusers] I have a problem trying use SIP

by Rogerio Agostinho

My friend, First I thank you for reading this e-mail and being interesting on my problem, I have SIP 0.8.10 installed in my Red Hat 7.2,, it was easy to install and configure the packages but after it all I’m not getting to connect to my SIP server. If I type the command “ser” I can see this message bellow: [root@afs sbin]# ser Listening on 127.0.0.1 [127.0.0.1]::5060 192.168.0.17 [192.168.0.17]::5060 Aliases: afs:5060 afs.advus.com:5060 localhost:5060 But when I try to check the port typing the command (telnet afs.advus.com 5060) just to prove the port 5060 is on the listening status and i just do not get any connection. I do not have a firewall installed on this machine and I m trying to connect inside my site, I mean, I am trying to connect inside my network not on the internet. I realy need to have the SIP server working, but I already did everything I knew to make it just works fine but as in all my trying I fail. Best Regards, Rogerio Agostinho --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.512 / Virus Database: 309 - Release Date: 8/19/2003

20 years, 8 months

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sr-users September 2003