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37551 discussions

[Serusers] Running SER Server behind NAT
by Alan 11 May '06

11 May '06

Is it possible to run SER SIP server behind a NAT? If so, what type of configuration changes am i looking at? My current scenario is as follows. Internet <-----> Pix (12.x.x.x translates to 10.x.x.x) <----> SIP Server Requests to and from the internet have a static NAT translation rule in a PIX firewall. Traffic flows from the outside global IP to an inside NAT'ed IP (non-routable) and vice versa. The SIP server's interface is configured with a non-routable IP addresss (10.x.x.x) for example. My problem is related to the message header and the ACK has a record route with the 10.x.x.x in it. Please help.

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[Serusers] Re: Serusers Digest, Vol 37, Issue 3 -- IM + Presence help
by Suraj Kumar 11 May '06

11 May '06

Hi Vaclav, Thanks for the help! That helped!! However, when I registered two users from windows messanger, I can login but I cannot see them with the serctl ul show command which shows the following message : >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> 500 Command 'ul_dump' not found >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> If i add a contact in the IM client, SER displays the following message: >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> subscription to invalid user sip:201@xxx.xxx.248.115 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> I have used the config sample from the example with DB in the presence hand book: but my user location is set as follows : modparam("usrloc", "db_mode", 0) Thanks a lot! Suraj

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[Users] problem in sending bye
by openser user 11 May '06

11 May '06

hi all, do any one faced a problem like, if i call a person and i will cut(disconnect) the call before it rings. even after cutting the call before it ring's the called person will get a ring. but when i cut the call after it rings it works properly(mean to say the called person gets disconnet before the called person pickup the call). with regards openseruser --------------------------------- Yahoo! India Answers: Share what you know. Learn something new. Click here Send instant messages to your online friends - NOW

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[Serusers] SER+asterisk MOH
by dashy dude 11 May '06

11 May '06

Dear All I am trying to deliver MOH service in the following manner; _____ _____ _____ |UA1 |<--->|SER |<--->|UA2| ______ _____ _____ | | ________ |Asterisk| ________ When UA1 is talking to UA2, either of them presses a hold button, and asterisk serves MOH Requesting all the gurus to comment on whether it is possible. and will appreciate if someone can guide what is needed on SER. Thanks D __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] Regarding on " radtest "
by raviprakash sunkara 11 May '06

11 May '06

Hi Everybody I'm working on openser + freeradius. Recently Installed both . I had a issue below onw, actual what this means... By Typing '" radtest 444 444 192.168.2.55 1812 radiustest " its shows below one. Sending Access-Request of id 236 to 192.168.2.55 port 1812 User-Name = "444" User-Password = "444" NAS-IP-Address = 255.255.255.255 Re-sending Access-Request of id 236 to 192.168.2.55 port 1812 User-Name = "444" User-Password = "444" NAS-IP-Address = 255.255.255.255 -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com

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[Serusers] Ser start but no PID
by Linh Pham 11 May '06

11 May '06

Hi, I compile SER 0.9.6 from source on a computer running Fedora Core 4. I suppose make all, make install all go well as I can't see any warning or error. When I start ser, it seems normal but I can't find the pid with ps. [root@localhost ~]# serctl start Starting SER : started pid(18726) [root@localhost ~]# ser Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.4 [192.168.1.4]:5060 udp: 192.168.1.17 [192.168.1.17]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.4 [192.168.1.4]:5060 tcp: 192.168.1.17 [192.168.1.17]:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: localhost:5060 udp: localhost.localdomain:5060 [root@localhost ~]# serctl stop Stopping SER : /usr/local/sbin/serctl: line 813: kill: (18726) - No such processstopped [root@localhost ~]# How can I fix this so ser run normally? Thank you. -- Best regards, Linh Pham

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[Users] Comma-Separated Via Header
by Dioris Moreno 10 May '06

10 May '06

I was making some fax tests using a Cisco AS5300, OpenSER 1.0.1 and a Grandstream HandyTone 486. Everything results fine when I previously make the same test directly from the HandyTone to the AS5300, however when I put OpenSER in the middle the fax connection could not be established. Checking captures at both ends, I noticed the HandyTone was not receiving the OK response to its RE-INVITE request, but the AS5300 was sending it. Looking a capture in the OpenSER, I suppose it didn't send the OK response from the AS5300 to the HandyTone because the format of the Via Header used by the AS5300. Look at the captures below, the AS5300 sends the Via header in a comma-separated format instead using a line for each one. Is there a way to workaround this? I think my best option is to modify the OpenSER code to support both formats. Is this something that somebody already does it? If the answer it is not, where can I start? I'll appreciate any help. Dioris Moreno Cisco AS5300 IP Address: AAA.BBB.CCC.DDD OpenSER IP Address: WWW.XXX.YYY.ZZZ Grandstream HT486 IP Address: FFF.GGG.HHH.III Calling Party Number (PSTN): UserA Called Party Number (VoIP): UserB Message From OpenSER to Cisco AS5300 Session Initiation Protocol Request-Line: INVITE sip:UserA@AAA.BBB.CCC.DDD:5060 SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:WWW.XXX.YYY.ZZZ;ftag=266f5f157860b9ca;lr=on> Via: SIP/2.0/UDP WWW.XXX.YYY.ZZZ;branch=z9hG4bK6a82.677fb8e1.0 Via: SIP/2.0/UDP FFF.GGG.HHH.III:50841;branch=z9hG4bK90436408698e6881 From: <sip:UserB@WWW.XXX.YYY.ZZZ>;tag=266f5f157860b9ca SIP from address: sip:UserB@WWW.XXX.YYY.ZZZ SIP tag: 266f5f157860b9ca To: <sip:UserA@AAA.BBB.CCC.DDD>;tag=D33F86B0-94F SIP to address: sip:UserA@AAA.BBB.CCC.DDD SIP tag: D33F86B0-94F Message From Cisco AS5300 to OpenSER Session Initiation Protocol Status-Line: SIP/2.0 200 OK Status-Code: 200 Resent Packet: False Message Header Via: SIP/2.0/UDP WWW.XXX.YYY.ZZZ;branch=z9hG4bK6a82.677fb8e1.0,SIP/2.0/UDP FFF.GGG.HHH.III:50841;branch=z9hG4bK90436408698e6881 From: <sip:UserB@WWW.XXX.YYY.ZZZ>;tag=266f5f157860b9ca SIP from address: sip:UserB@WWW.XXX.YYY.ZZZ SIP tag: 266f5f157860b9ca To: <sip:UserA@AAA.BBB.CCC.DDD>;tag=D33F86B0-94F SIP to address: sip:UserA@AAA.BBB.CCC.DDD SIP tag: D33F86B0-94F

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[Users] Stateful Transfers
by Douglas Garstang 10 May '06

10 May '06

I got a big problem here that I don't know how to fix. An OpenSER system forwards calls from phones to Asterisk boxes. Up until now we had been using the dispatcher to evenly distribute calls to the Asterisk boxes. However, when someone transfers a call, using the dispatcher, the call can go to a different Asterisk box, which breaks the transfer. I need to find some way to make OpenSER route transfers to the same Asterisk system. I'm not sure how to do this. If there was some way to, upon an INVITE, check and see if there was already a call in progress for that user and then send it to the same Asterisk system. Would stateful processing help in this way? Can stateful processing 'remember' that there's a call already in progress like this, or does the stateful simply relate to stateful processing of a single call from INVITE to OK/ACK? Doug.

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[Users] failure_route
by D'Addelfio Davide 10 May '06

10 May '06

Hi all, i have 2 questions for the forum: 1) What can I do to simulate a failure message to understand if openser store it into database, through acc_db_request? 2) Have I any chance to get information from reply messages, like storing some in db or something else? I know I can't use acc and avp module in on_reply route block, so any suggestion? Thanks in advance Davide

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[Users] Call routing
by Jeff Williams 10 May '06

10 May '06

This is not openser specific, but can I just check my understanding of the SIP RFC? Given an INVITE where the Request-URI is different from the To field, a UAS should route the INVITE based on the Request-URI, correct? The reason I ask it that I am having problems integrating openser with a proprietry SIP proxy as the proprietry proxy routes INVITES based on the To field. Jeff

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