February 2004 - sr-users

[Serusers] SIP/2.0 483 Too Many Hops ??
by Nair 21 Feb '04

21 Feb '04

hi all, I am trying to use SER as below: UA1>>>>>>>>>B2BUA>>>>>>>>SER------>GW UA1:192.168.10.198 B2BUA:192.168.10.144 SER:192.168.10.1 GW:192.168.10.156 When i try to make calls from UA1 i am getting mesg "SIP/2.0 483 Too Many Hops" what cud be wrong?? regards manu. --------------------------------- Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want.

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[Serusers] Is STUN protocol usable behind firewall?
by wangji 21 Feb '04

21 Feb '04

Hi all, In my test environment there are a firewall between SER server and SIP terminal(But I am not sure). I think so because in this envirmonent SJPhone can not register to SER server (I use ethereal on SER server and can't find register packets from SJPHONE). I had tested SJPHONE behind a NAT created by cisco router and success. So I think that thing is a firewall. Today I test X-Lite in the same environment with same setting but it works. I get the UDP packet by ethereal. I find it send packet to xtunnel.xten.net port 3478/3479. These ports are used by STUN protocol. I would why it works. Attached file can be open by ethereal. In the file no reponse register is sent by SJPhone and those success is sent by X-Lite Br, Wangji

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[Serusers] SIP iptables question
by Dee Lowndes 20 Feb '04

20 Feb '04

I have finally worked out the problem to an iptables issue and was wondering if anyone had some rules to sort this out. If two clients are on an external IP it works, if two clients are on an internal IP it works but if one client is internal and the other is external then the external guy can hear the internal but the internal guy hears nowt. Any idea/modules/rules to get around this? Thanks in advance, Dee

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[Serusers] Proxy requests between domains
by Alan Litster 20 Feb '04

20 Feb '04

Hi all, I've got a question regarding proxying requests between domains, I hope someone can help me... Take for example two domains, reseller.com & gateway.com. A SUA hangs off of reseller.com and places a call to the PSTN. For it to go out via the PSTN reseller.com forwards it onto gateway.com. Is there some kind of check you can do in ser.cfg that would allow you to determin if the request came from the reseller.com proxy or directly from the SUA? I'd like to stop SUA's from directly placing calls out via gateway.com and forcing them to go via there own sip servers. Then using this method along with RADIUS authentication I think it should be possible to secure our gateways? One of my colleagues suggested writing a module that would compare the given uri against a username/IP/domain in a database....? Many Thanks, Alan ------------------------------------------------------------------------------------------------------- This email, and any files transmitted with it, is copyright and may contain confidential information. The contents are intended for the use of the addressee(s) only. Unauthorized use may be unlawful. If you receive this email by mistake, please advise sender immediately. The views of the author may not necessarily constitute the views of Telco Electronics Limited. Nothing in this mail shall bind Telco Electronics Limited in any contract or obligation. Telco Electronics Limited 6-8 Oxford Court Brackley Northants NN13 7XY Tel 07000 701999 Fax 07000 701777

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[Serusers] Strange problems
by Dee Lowndes 20 Feb '04

20 Feb '04

I have ser working over local ip and sort of working externally but it sounds Dr Who Darlek like over the external link and only one side has audio. Could this be a firewall issue? Dee

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[Serusers] Serweb problem: Voicemail store greeting failed
by Raphael Coeffic 20 Feb '04

20 Feb '04

Do you mean that you don't get serweb to upload your greetings? If yes, it has nothing to do with SEMS. Your problem is only related to the PHP code... You should take a look at your serweb configuration again. You could also try to enable PHP debug in the php configuration. -Raphael. ----- Original Message ----- From: "Dee Lowndes" <dee(a)asyouneed.com> To: <sems(a)lists.iptel.org> Cc: <rco(a)iptel.org> Sent: Friday, February 20, 2004 5:29 PM Subject: RE: [Serusers] Voicemail store greeting failed > Hi Raphael, > > Thanks for the pointer I got the feeling that it was slightly > out of date I now have sems showing up under ps aux but I am still > getting the store greeting failed message. > > I thought it might be a folder permission thing but changing > /usr/local/lib/sems/audio to the apache user but it didn't sort it. > > Any ideas? > > Thanks, > Dee > > -----Original Message----- > > From: Raphael Coeffic [mailto:rco@iptel.org] > > Sent: 20 February 2004 15:39 > > To: Dee Lowndes; serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Voicemail store greeting failed > > > > Hi, > > > > The documentation you are pointing at is outdated. Please follow the > > instructions in the README file present in the SEMS source tree. > > oRtp is no more needed as SEMS now has its own RTP stack. You can find > > more > > informations at http://sems.berlios.de. > > If it still doesn't work, please contact us at sems(a)lists.iptel.org. > > > > -Raphael. > > > > ----- Original Message ----- > > From: "Dee Lowndes" <dee(a)asyouneed.com> > > To: <serusers(a)lists.iptel.org> > > Sent: Friday, February 20, 2004 11:41 AM > > Subject: [Serusers] Voicemail store greeting failed > > > > > > > Hi Again, > > > > > > I am trying to get voicemail to work but I keep store greeting > > > failed. I have downloaded ortp-0.6.2 and installed it and made > changes > > > to my ser.cfg file as detailed > > > http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1318 > > > > > > Still know joy any ideas? > > > > > > Thanks, > > > Dee > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Is STUN protocol usable behind firewall?
by wangji 20 Feb '04

20 Feb '04

Hi all, In my test environment there are a firewall between SER server and SIP terminal(But I am not sure). I think so because in this envirmonent SJPhone can not register to SER server (I use ethereal on SER server and can't find register packets from SJPHONE). I had tested SJPHONE behind a NAT created by cisco router and success. So I think that thing is a firewall. Today I test X-Lite in the same environment with same setting but it works. I get the UDP packet by ethereal. I find it send packet to xtunnel.xten.net port 3478/3479. These ports are used by STUN protocol. I would why it works. Attached file can be open by ethereal. In the file no reponse register is sent by SJPhone and those success is sent by X-Lite. The file is bigger than 40K so I compress it by WinRAR. I trying X-Lite again and it says the firewall is port restricted single mapped port system. What's that means? Br, Wangji

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[Serusers] Sometimes registrar module's desc_time_order options fail
by wangji 20 Feb '04

20 Feb '04

Hi all, My network is like this: ---------------------------- | | | | | Private network | ----------- NAT----------Internet ----------------- SER server | | | | ---------------------------- And I use SER+mysql DB. For a SIP terminal there are only one number and for a number there are only one SIP terminal. For a call I need to know only one URI. So that I set : modparam("registrar", "append_branches", 0) That mean I must get the last register infomation for a SIP terminal because of NAT port maybe change when SIP terninal reboot(Maby power off and power on so that maybe no unregister packets sent). So that I set: modparam("registrar", "desc_time_order", 1) to get the most current URI. I don't know what mean with the parm default_q, sometimes I set it as 50, sometimes use default value 0. But SER often call to the URI wrong. Looking my recorder: I use a SIP terminal named 1001. The terminal register multi times. So I can get many I get usrloc information by command "serctl ul show" like this: ===Domain list=== ---Domain--- name : 'location' size : 512 table: 0x402e7998 d_ll { n : 2 first: 0x402e99a0 last : 0x402e9a80 } ...Record(0x402e99a0)... domain: 'location' aor : '1001' ~~~Contact(0x402e9b48)~~~ domain : 'location' aor : '1001' Contact: 'sip:1001@218.107.145.102:7035' Expires: 131 q : 0.00 Call-ID: '3BAFC06D5BD54939A3114CABF35D1FE0(a)novsky.com' CSeq : 11838 replic : 0 State : CS_SYNC Flags : 1 next : 0x402e9be8 prev : (nil) ~~~/Contact~~~~ ~~~Contact(0x402e9be8)~~~ domain : 'location' aor : '1001' Contact: 'sip:1001@218.107.145.102:7011' Expires: 36 q : 0.00 Call-ID: 'D125943999FF4907BCD175EEFE3DE24A(a)novsky.com' CSeq : 43236 replic : 0 State : CS_SYNC Flags : 1 next : 0x402e99e0 prev : 0x402e9b48 ~~~/Contact~~~~ ~~~Contact(0x402e99e0)~~~ domain : 'location' aor : '1001' Contact: 'sip:1001@218.107.145.102:7080' Expires: 1794 q : 0.00 Call-ID: '9658F25C5E3040C1AE4C3405032B7558(a)novsky.com' CSeq : 63264 replic : 0 State : CS_NEW Flags : 1 next : (nil) prev : 0x402e9be8 ~~~/Contact~~~~ .../Record... ...Record(0x402e9a80)... domain: 'location' aor : '1000' ~~~Contact(0x402e9ac0)~~~ domain : 'location' aor : '1000' Contact: 'sip:1000@218.201.88.163:1083' Expires: 2565 q : 0.00 Call-ID: '3350078894(a)172.16.3.80' CSeq : 4 replic : 0 State : CS_SYNC Flags : 1 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... In the list we can find that last register use NAT port 7080 by check expires time. I use this terminal -- 1001 to call 1001, It should return 486 BUSY in normal. But it didn't reponse. See that packet I get by tethereal: 291.677702 218.107.145.102 -> 218.201.88.164 SIP/SDP Request: INVITE sip:1001@novsky.com, with session description 291.702534 218.201.88.164 -> 218.107.145.102 SIP Status: 407 Proxy Authentication Required 291.822811 218.107.145.102 -> 218.201.88.164 SIP Request: ACK sip:1001@novsky.com 291.824597 218.107.145.102 -> 218.201.88.164 SIP/SDP Request: INVITE sip:1001@novsky.com, with session description 291.868197 218.201.88.164 -> 218.107.145.102 SIP Status: 100 trying -- your call is important to us 291.870300 218.201.88.164 -> 218.107.145.102 SIP/SDP Request: INVITE sip:1001@218.107.145.102:7035, with session description 292.045669 218.201.88.164 -> 218.107.145.102 SIP/SDP Request: INVITE sip:1001@218.107.145.102:7035, with session description 294.065633 218.201.88.164 -> 218.107.145.102 SIP/SDP Request: INVITE sip:1001@218.107.145.102:7035, with session description 298.105648 218.201.88.164 -> 218.107.145.102 SIP/SDP Request: INVITE sip:1001@218.107.145.102:7035, with session description I find SER get the wrong URI and send to NAT port 7035. It sometimes happens but not always. How to deal with that. Br, Wangji

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[Serusers] Voicemail store greeting failed
by Dee Lowndes 20 Feb '04

20 Feb '04

Hi Again, I am trying to get voicemail to work but I keep store greeting failed. I have downloaded ortp-0.6.2 and installed it and made changes to my ser.cfg file as detailed http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1318 Still know joy any ideas? Thanks, Dee

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[Serusers] Questions about Serweb
by Mike Fallows 20 Feb '04

20 Feb '04

Got my ser server working and have installed serweb. Everything seems to be fine. One slight problem is that when I do a find user from the user_interface/index.php I get a runtime error on line 134. Doing the same thing from the admin interface is fine. Any ideas? I have also noticed that each user receives the same alias number when signing up ($this->first_alias_number=82000;). Can this be incremented automatically by the sign up? Thanks in advance. Mike

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