sr-users September 2007

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158 participants
248 discussions

Re: [OpenSER-Users] Missing RTP stream
by Morten Isaksen 21 Sep '07

21 Sep '07

The error was in Mediaproxy. In rtphandler.py i changed nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '172.16.0.0', 'value': 0xac100000L, 'mask': 0xfff00000L}, {'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask': 0xffff0000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] To nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] I think Mediaproxy got confused with the RFC1918 IP's. In my setup there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a router. On 9/21/07, Norman Brandinger <norm(a)goes.com> wrote: > Is there a firewall in the picture ? You have two different subnets and > there probably is a box doing some (NAT) translation / routing between > them. Is is possible the RTP stream is being blocked at the firewall ? > > Norm > > > Morten Isaksen wrote: > > Hi! > > > > I can see in the mediaproxy log the it is initialized to proxy the > > call, but I newer get a "session xxxxx: called signed in from xxx" > > from Asterisk. > > > > session.py shows that the the connection between mediaproxy and > > Asterisk is missing. > > > > I will try to take a look at the sip debug from asterisk and try to > > change the NAT settings in Asterisk. > > > > Thanks for your input. > > > > On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > > > >> You stated that you've forced every call through mediaproxy. Are you > >> positive ? > >> > >> Have you taken a look at the mediaproxy logs (and/or sessions.py when > >> the call is up) ? They might provide some useful information to you. > >> > >> Ditto for Asterisk "sip set debug on" (note that the sip debug command > >> format is a moving target). > >> > >> Have you looked at the "nat=" settings in sip.conf as well ? At times, > >> they tie closely with "canreinvite=". > >> > >> Norm > >> > >> > >> Morten Isaksen wrote: > >> > >>> Hi! > >>> > >>> canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the > >>> clients IP-addresses from Asterisk, so I am pretty sure that this is > >>> not the issue. > >>> > >>> On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > >>> > >>> > >>>> Hi Morten, > >>>> > >>>> Admittedly, I haven't looked closely at your trace. However, based on > >>>> the description you gave, the first place to look is at the "canrevite" > >>>> setting in Asterisk sip.conf. You might want to try "canreinvite=no" > >>>> after reading up on this particular setting. > >>>> > >>>> Regards, > >>>> Norm > >>>> > >>>> > >>>> Morten Isaksen wrote: > >>>> > >>>> > >>>>> Hi! > >>>>> > >>>>> I have a strange problem with a missing RTP stream between OpenSER and > >>>>> Asterisk. I am not sure if it is OpenSER og Asterisk related. > >>>>> > >>>>> I have this setup > >>>>> > >>>>> Phone A (172.17.96.17) -- > >>>>> \ Openser -- Asterisk > >>>>> -- PSTN > >>>>> / (192.168.0.6) (192.168.0.3) > >>>>> Phone B (172.17.96.10) -- (172.17.64.1) > >>>>> > >>>>> I also have a Mediaproxy running on OpenSER and I force every call to > >>>>> use the Mediaproxy. > >>>>> > >>>>> I call from Phone A or B to the PSTN works fine and from PSTN to Phone > >>>>> A or B it also works. > >>>>> > >>>>> I have the dialplan logic on my Asterisk server so I want calls from > >>>>> Phone A to Phone B to pass the Asterisk server. And this is were I > >>>>> have the problem. When the call is established the RTP stream is > >>>>> missing between Mediaproxy and Asterisk. I only have a RTP stream > >>>>> between the phones and Mediaproxy. As far as I can see the SIP > >>>>> signalling is correct. > >>>>> > >>>>> The SIP traces is listed below. Can you spot the problem in this? > >>>>> > >>>>> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can > >>>>> help me solve this problem. > >>>>> > >>>>> SIP trace between the phones and OpenSER: > >>>>> > >>>>> > >>>>> > >>> > >>> > >> > > > > > > > > -- Morten Isaksen http://www.misak.dk/blog/

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[OpenSER-Users] Sending IM even when the destination User goes "Offline"
by Cesar Troncoso 21 Sep '07

21 Sep '07

Hello everybody, Lets suppose a user wants to send an Insant Message to another user of his contact list whose presence is "offline". Is there a way to store the message somewhere (another server maybe) for a certain period of time, so that when the destination user goes on-line again receives the Instant Message? Thank you very much in advanced for your help, Cesar

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[OpenSER-Users] Authentication on SIP Invite Message
by Patrick Miccio 21 Sep '07

21 Sep '07

hi, you should have a look at the OpenSER modules like AUTH / AUTH_DB http://www.openser.org/docs/modules/ cheers, Patrick

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[OpenSER-Users] Any way to cache SRV record lookups?
by Tim Madorma 21 Sep '07

21 Sep '07

Hi, I have an application where I am using the OpenSER with an SRV record lookup to load-balance towards several destination servers. When testing, I noticed that even if I have nscd caching turned on by setting: enable-cache hosts yes in my nscd.conf file, it still sends out a DNS SRV query for each call that is made (note that for A record lookups it does use the local nscd cache instead of doing a DNS query). I am running my host on Solaris. Is there any way to get nscd to cache SRV records? If not is there another alternative that I can use? thanks, Tim

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[OpenSER-Users] Authentication on SIP Invite Message
by Memon,Arshad Mehrab 21 Sep '07

21 Sep '07

Hi, Is there any way to enforce security on SIP invite Messages, The SIP proxy should send 407 (Proxy Authentication Required) in response to first INVITE. -- Best Regards, Arshad

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[Serusers] about "calls_forwarding" table in SER database
by 作仁 21 Sep '07

21 Sep '07

Hi all Recently, I try to know how to use "calls_forwarding" table I had already search the document on the iptel.org website But, I can't find anything information about this table. Any one can tell me what value should stuff into this table? Thanks.

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[Serusers] New to SIP Express Server
by Do Nguyen Ha 21 Sep '07

21 Sep '07

hi all the SIP Express Server is a new thing to me Can anyone reccommend me where to start to understand & to config & to install the SIP Express Server + SEMS Thanks Do Nguyen Ha

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Re: [OpenSER-Users] creating call number for xlite user
by Live Great 21 Sep '07

21 Sep '07

Thanks for the info. In the openser server, can you tell me how to create phone number that allow xlite user to dial in (for testing purpose)? Thanks S ----- Original Message ---- From: Norman Brandinger <norm(a)goes.com> To: Live Great <livegreat007(a)yahoo.com> Cc: users(a)openser.org Sent: Friday, September 21, 2007 11:16:31 AM Subject: Re: [OpenSER-Users] creating call number for xlite user Executing openserctl without any parameters will display some usage information. You probably want to start out adding a user to the subscriber table. They syntax is shown below: openserctl add <username> <password> <email> rpid is the Remote-Party-ID and is a SIP header. This is defined in the following EXPIRED draft: http://tools.ietf.org/html/draft-ietf-sip-privacy-04 There are others on this list that might know what the current status of this header is. You may also want to research the P-Asserted-Identity (RFC3325) SIP header. There is a lot of documentation on the openser.org web site. Regards, Norm Live Great wrote: > Hi, > > I have completed the configuration of openser in freebsd 6.2 and > downloaded xlite to my windows xp. > In the man page of opernserctl, I am not fully understand how to add a > calling number for remote xlite user. > eg. what is rpid? the man page doesn't explain. > > Can anyone help? > > Thanks > Sam > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] creating call number for xlite user
by Live Great 21 Sep '07

21 Sep '07

Hi, I have completed the configuration of openser in freebsd 6.2 and downloaded xlite to my windows xp. In the man page of opernserctl, I am not fully understand how to add a calling number for remote xlite user. eg. what is rpid? the man page doesn't explain. Can anyone help? Thanks Sam

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[OpenSER-Users] Asterisk + OpenSER + WeSIP + Cisco Call Manager
by Victor Pascual Ávila 20 Sep '07

20 Sep '07

Hello, we're deploying a new ToIP network and we are currently using Asterisk (for some services), and OpenSER (as a trunk). Now we would like to introduce a Cisco Call Manager (registered to the OpenSER) and a WeSIP. Do you have any experience connecting CCM (using SIP) to OpenSER? What about WeSIP interaction with other different proxies (not OpenSER+SEAS)? Thanks in advance, Victor Pascual

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sr-users September 2007