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11 participants
37462 discussions

[Serusers] Serweb won't authenticate
by Gavin Bensom 04 Sep '03

04 Sep '03

There was a similar post on Aug 13th. I'm having the same problem [http://lists.iptel.org/pipermail/serusers/2003-August/002259.html] and I've already tried the suggested solutions. Basically, on the user login or the admin login of serweb, when I type a username and password and press login or <enter> the page blanks, reloads to the login page, and does nothing. I get no errors of any kind. I have register_globals=On in php.ini. I'm relatively sure my php.ini is in the correct place [i.e. is being used by the server] How would I check this? My subscriber table in mysql has the correct domain in the domain column for all my users, including admin. I believe that I've made all the changes in Dan Austins HOWTO for serweb/php configuration. RedHat 9.0 kernel 2.4.20-20smp ser 0.8.11 (i386/linux) main.c, v 1.162.2.5 PHP 4.3.1 Apache 2.0.40 Any ideas? Thanks, G. --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.

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[Serusers] one way RTP session
by Samy Touati 04 Sep '03

04 Sep '03

Hi, Just to add on my previous posting, where the snom phone didn't have an RTP stream toward the other end, after doing some more tests I found that by chaniging who initiates the call I still get a one-way RTP stream. What can trigger such a behaviour where a device won't initiate an RTP stream after receiving an SDP. Thanks. Samy.

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[Serusers] Radius logon problems
by Rork, Joseph (J.P.) 04 Sep '03

04 Sep '03

I am trying to get my setup working with radius. I have the lastest freeradius, radiusclient, and ser. The test programs that come with radiusclient work fine, so I've determined that the problem now comes down to my ser setup. I've attached my ser.cfg as well as the output of ser debug. Any ideas? <<attempt>> <<ser.cfg>> Regards, Joseph Rork Ford Motor Company Real-Time Collaborative Applications, SIE Phone: 313-594-6672 Email: jrork(a)ford.com "There are 10 types of people in the world: Those who understand binary, and those who don't."

1 0

[Serusers] Script location
by Iván Ramírez 04 Sep '03

04 Sep '03

Hello all, I want to make a forward to another uri when something happens.. I saw an script example at the ser site... and I found the script I need.. but I don't know where I have to write the script...? on a separate file... on the config file...? where... Thanx in advance Ivan

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[Serusers] Online limit
by giAndrea 04 Sep '03

04 Sep '03

Hi, is possible that MSN Messneger limit my online user? I can see only 2 user online. thanks. Andera ---------------------------------- Legge di Clarke sulle idee rivoluzionarie Ogni idea rivoluzionaria - in campo scientifico, artistico o altro provoca tre stadi di reazione, riassumibili nelle seguenti frasi: 1. "E' impossibile; non farmi perdere tempo". 2. "E' possibile, ma non val la pena di farlo". 3. "L'ho sempre detto, io, che era un'ottima idea".

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[Serusers] bad request from asterisk to ser 0.8.11
by Kelvin Chua 04 Sep '03

04 Sep '03

encountered bad request for calls from asterisk to ser. i can't figure out why? does ser expect a "user" or port in To: field? failed codec negotiation? here are the sip messages Reliably Transmitting: INVITE sip:kelvin@10.17.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf From: "2010" <sip:2010@10.17.0.100:5065>;tag=as0dd619ef To: <sip:kelvin@10.17.0.100> Contact: <sip:2010@10.17.0.100:5065> Call-ID: 20be6f95408969da5117a6b82036d540(a)10.17.0.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 234 v=0 o=root 20301 20301 IN IP4 10.17.0.100 s=session c=IN IP4 10.17.0.100 t=0 0 m=audio 13602 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 10.17.0.100:5060 Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf From: "2010" <sip:2010@10.17.0.100:5065>;tag=as0dd619ef To: <sip:kelvin@10.17.0.100> Call-ID: 20be6f95408969da5117a6b82036d540(a)10.17.0.100 CSeq: 102 INVITE Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 Warning: 392 10.17.0.100:5060 "Noisy feedback tells: pid=20118 req_src_ip=10.17.0.100 req_src_port=5065 in_uri=sip:kelvin@10.17.0.100 out_uri=sip:10.17.0.30:12532;transport=tcp via_cnt==1" 9 headers, 0 lines Sip read: SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf From: "2010" <sip:2010@10.17.0.100:5065>;tag=as0dd619ef To: <sip:kelvin@10.17.0.100>;tag=49dd952c-6360-4555-be16-601328c6805a Call-ID: 20be6f95408969da5117a6b82036d540(a)10.17.0.100 CSeq: 102 INVITE Record-Route: <sip:kelvin@10.17.0.100;transport=tcp;r2=on;ftag=as0dd619ef;lr> Record-Route: <sip:kelvin@10.17.0.100;r2=on;ftag=as0dd619ef;lr> User-Agent: Windows RTC/1.0 Content-Length: 0 10 headers, 0 lines Transmitting: ACK sip:kelvin@10.17.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf From: "2010" <sip:2010@10.17.0.100:5065>;tag=as0dd619ef To: <sip:kelvin@10.17.0.100>;tag=49dd952c-6360-4555-be16-601328c6805a Contact: <sip:2010@10.17.0.100:5065> Call-ID: 20be6f95408969da5117a6b82036d540(a)10.17.0.100 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0

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[Serusers] snom 100 & SER strange behaviour
by Samy Touati 04 Sep '03

04 Sep '03

Hi, I don't know if this is the right place to post this, but I noticed a strange behaviour with the snom 100 with ser 0.8.11 and a grandstream. The snom is behind NAT, but ser is on a public address, the grandstream is on another network also behind nat. The nat is sip aware (cisco soho91). When I place a call from the snom to another IP phone, grandstream, the call doesn't always complete successfully: in 50% of cases after the other end answers the call, I can hear the other party, but they cannot hear me. When I look into ethereal, I find the regular SIP INVITE and rining messages, then I see a bunch of UDP packets (the RTP session) from the other end, the grandstream, but I don't see a stream of RTP from the snom to the other end. Now what is strange here is that sometimes everything works fine, and we can both hear each others, and I do then see the RTP stream in both directions. Both devices share the same codec, and the SDP info is ok, the ports numbers are valid. I don't think this is ser related, but I wonder if somebody has experienced such a behaviour. Thanks. Samy.

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Re: [Serusers] Accounting and parallel/sequential forking
by Jiri Kuthan 03 Sep '03

03 Sep '03

At 12:02 AM 9/4/2003, Guilherme Dal Pizzol wrote: >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? That's correct -- there is only one final result for a proxied transaction (no matter how many branches you fork) and that's the value forwarded upstream. Per 3261, that's module the lowest error code. -jiri

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[Serusers] Accounting and parallel/sequential forking
by Guilherme Dal Pizzol 03 Sep '03

03 Sep '03

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.

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[Serusers] :: nathelper vs jasomi
by abdoul 03 Sep '03

03 Sep '03

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

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