sr-users July 2004

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178 participants
449 discussions

[Serusers] sip softphone
by varala ramakanth 21 Jul '04

21 Jul '04

hello friends, does we have any free windows based softphones with g723 codec which run with sip express router with regards rama kanth __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

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RE: [Serusers] Problem with messages + msilo
by Dave Bath 20 Jul '04

20 Jul '04

(Sorry if this is reposted, but noticed I had some failure reports for this, so not sure if it made it to the mailing list or not) Hey there Andrei (or anyone else!), Here are the syslog messages and ngrep for a serweb send message. Serweb reports message sent ok, but I don't receive it on the client, nor is it stored in the Message Store? Any ideas would be very welcome! Dave MSILO: the downstream UA does not support MESSAGE requests ... Jul 20 08:49:49 sip /usr/sbin/ser[23058]: ACC: transaction answered: method=MESSAGE, i-uri=sip:admin@161.30.94.68, o-uri=sip:admin@161.30.94.70:5060, call_id=73f9fea9-0(a)127.0.0.1, from=sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54ed -fba5, code=202 Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ERROR: udp_send: sendto(sock,0xbd777de0,667,0,0xbd779394,16): Invalid argument(22) Jul 20 08:49:52 sip /usr/sbin/ser[23017]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Jul 20 08:49:52 sip /usr/sbin/ser[23017]: msg_send: ERROR: udp_send failed Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ERROR: t_forward_nonack: sending request failed Jul 20 08:49:52 sip /usr/sbin/ser[23017]: ACC: transaction answered: method=MESSAGE, i-uri=sip:daemon@mydomain.org, o-uri=sip:daemon@mydomain.org, call_id=73f9fea8-0(a)161.30.94.68, from=sip:sip_registrar@161.30.94.68;tag=533cb9e91f4b999cf76861cbb9ed54ed -1247, code=477 ngrep UDP port 5060 interface: eth0 (161.30.94.64/255.255.255.224) filter: ip and ( port 5060 ) match: UDP ########################### U 161.30.94.68:5060 -> 161.30.94.70:5060 MESSAGE sip:admin@161.30.94.70:5060 SIP/2.0..Max-Forwards: 10..Record-Route : <sip:admin@161.30.94.68;ftag=533cb9e91f4b999cf76861cbb9ed54ed-fba5;lr=on > ..Via: SIP/2.0/UDP 161.30.94.68;branch=z9hG4bK33b2.acca75b7.0..Via: SIP/2.0 /UDP 161.30.94.68;branch=z9hG4bKb789.54d3c0e.0..To: <sip:admin@161.30.94.68 >..From: sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54e d-fba5..CSeq: 10 MESSAGE..Call-ID: 73f9fea9-0@127.0.0.1..Content-Length: 16 ..User-Agent: Sip EXpress router(0.8.12 (i386/linux))..p-version: Web_inter face_Karel_Kozlik-0.9..Contact: <sip:daemon@mydomain.org>..Content-Type: te xt/plain; charset=UTF-8....test message.... ## U 161.30.94.70:5060 -> 161.30.94.68:5060 SIP/2.0 405 Method Not Allowed..Via: SIP/2.0/UDP 161.30.94.68;branch=z9hG4b K33b2.acca75b7.0..Via: SIP/2.0/UDP 161.30.94.68;branch=z9hG4bKb789.54d3c0e. 0..From: sip:test1@sip.dev.inmarsat.com;tag=533cb9e91f4b999cf76861cbb9ed54e d-fba5..To: <sip:admin@161.30.94.68>;tag=3946696931..Contact: <sip:admin@16 .30.94.70:5060>..Call-ID: 73f9fea9-0@127.0.0.1..Allow: INVITE,ACK,BYE,CANC EL,OPTIONS,NOTIFY..CSeq: 10 MESSAGE..Server: X-Lite release 103m..Content- Length: 0.... #exit 31 received, 0 dropped -----Original Message----- From: Andrei Pelinescu-Onciul [mailto:pelinescu-onciul@fokus.fraunhofer.de] Sent: 19 July 2004 15:37 To: Dave Bath Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Problem with messages + msilo On Jul 19, 2004 at 13:32, Dave Bath <dave(a)fuuz.com> wrote: > Hey there, > > Once again, many thanks for a swift response Andrei! No, the machine > does not have two interfaces, it's a single public ip address. What do > you mean about ingress filtering? Which network dumps would be helpful? If it's a single ip, than don't worry about ingress filtering. Just dump the udp traffic to your machine port 5060 until you get one of the log error message (sendto ...) and then send use the log and the dump. The message might also be caused by some UA using a 0 source port (which is highly broken), or a 0 destination port or ip in the message uri. Andrei

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[Serusers] run rtpproxy as non root
by Richard 20 Jul '04

20 Jul '04

Hi, When i tried to run rtpproxy as non root, it gave an error "rtpproxy: can't bind to a socket: Permission denied". Is there a reason to run it as root? Thanks, Richard __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

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[Serusers] How to configure SER for that kind of use
by l.gaillard01＠free.fr 20 Jul '04

20 Jul '04

Hi all, I need to configure two ser servers to do such a thing : SER2 is the registrar proxy, knowing ips ... SER1 is a must go through proxy. Each packet must go through him before being sent to SER2. ________ ________ | | REGISTER | | Phone B ---->| |------------------>| | | SER1 | | SER2 | | | | | | IP .4 | REGISTER | IP .8 | Phone A ---->| |------------------>| | |_______ | |________| CALL from A to B ________ ________ | | CALL | | Phone B <----| |<------------------| | | SER1 | | SER2 | | | | | | IP .4 | CALL | IP .8 | Phone A ---->| |------------------>| | |_______ | |________| When a call is done by A, it must go through ser1 then to ser2 and before coming back to phone B it must go to SER1. I can't find a way to do that with a script. If someone can help me, it would be great. Thanks, Laurent

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[Serusers] nat question
by Shi Hoch 20 Jul '04

20 Jul '04

I'm building a system that "sits" both in the LAN and in the Wan. My question is this: Is there a way to check the IP address of the person I'm trying to contact? So when a LAN user contacts a LAN user or WAN (not behind Nat) user contacts a WAN (not behind Nat) user They will be no use of the RTP proxy? Something like: if (contact_ip_addr != src_ip ) { ......}; Thanx Shi Hoch

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RE: [Serusers] NAT vs. NoNat authentication
by dhiraj.2.bhuyan＠bt.com 20 Jul '04

20 Jul '04

There are 3 possible points where the private IP might have been replaced with the ADSL gateway's public address. 1. On the client itself (using STUN?) 2. On the ADSL gateway (Is it SIP aware?) 3. Misconfigured SER (I think this is not the case - since it works for port 5070). To confirm if it is (1) - run a sniffer like ethereal or tcpdump to capture packets as it leaves the client machine - both for port 5060 and 5070. See if the client is doing anything smart - replacing private IP with gateway IP while using 5060 and not for 5070? To confirm if it is (2) - run a sniffer on the same collission domain or on the same machine where SER is running and capture the registration request packets. If (1) is false and still the private IP is getting modified, its the ADSL gateway that's doing some proxying of SIP traffic (port 5060). Which ADSL gateway are you using? Dhiraj -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Bart Van Daal Sent: 20 July 2004 09:16 To: serusers(a)lists.iptel.org Subject: RE: [Serusers] NAT vs. NoNat authentication Hello I'll post the answers to the two replies: >Andrei> Do you have another UA behind the same nat, using 5060? No it's only 1 Phone ---- ADSLRouter(NAT) ------- Internet ---------- Ser Dhiraj > ..i'l post the two ngreps again: I'm sorry for the long post. What I can see is, when ser runs on port 5070 the register contains a private ip in the 'Via:' header. When it runs on 5060 The 'Via:' header contains the public IP of the router and an unprivileged port. ----------------------------------------------- 5070 ----------------------------------------------- filter: ip and ( port 5070 ) # U 213.219.137.137:5070 -> 212.71.0.60:5070 REGISTER sip:ser.edpnet.net:5070 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.2:5070. Supported: replaces. User-Agent: SIP201 (lp201sip.100a). Contact: <sip:bart@10.0.0.2:5070>;expires=60. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-0-42e-7fea. To: <sip:bart@ser.edpnet.net>. Call-ID: a000002-13ce-0-406-79bf-1. CSeq: 1 REGISTER. Content-Length:0. . # U 212.71.0.60:5070 -> 213.219.137.137:5070 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.0.0.2:5070;rport=5070;received=213.219.137.137. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-0-42e-7fea. To: <sip:bart@ser.edpnet.net>;tag=2497a39c629b119dac83769f58cd2b29.1cd2. Call-ID: a000002-13ce-0-406-79bf-1. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="ser.edpnet.net", nonce="40fcce4ec4ab3796c95cb2c87a9d94a05651ed08". Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)). Content-Length: 0. Warning: 392 212.71.0.60:5070 "Noisy feedback tells: pid=18743 req_src_ip=213.219.137.137 req_src_port=5070 in_uri=sip:ser.edpnet.net:5070 out_uri=sip:ser.edpnet.net:5070 via_cnt==1". . # U 213.219.137.137:5070 -> 212.71.0.60:5070 REGISTER sip:ser.edpnet.net:5070 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.2:5070. Supported: replaces. User-Agent: SIP201 (lp201sip.100a). Contact: <sip:bart@10.0.0.2:5070>;expires=60. Authorization: Digest username="bart", realm="ser.edpnet.net", nonce="40fcce4ec4ab3796c95cb2c87a9d94a05651ed08", uri="sip:ser.edpnet.net:5070", response="ea0329c8f3a4d199230733feb750d3a1", algorithm=MD5. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-40fccd8d-1991-7051. To: <sip:bart@ser.edpnet.net>. Call-ID: a000002-13ce-0-406-79bf-1. CSeq: 2 REGISTER. Content-Length:0. . # U 212.71.0.60:5070 -> 213.219.137.137:5070 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.2:5070;rport=5070;received=213.219.137.137. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13ce-40fccd8d-1991-7051. To: <sip:bart@ser.edpnet.net>;tag=2497a39c629b119dac83769f58cd2b29.1cd2. Call-ID: a000002-13ce-0-406-79bf-1. CSeq: 2 REGISTER. Contact: <sip:bart@213.219.137.137:5070>;expires=60. Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)). Content-Length: 0. Warning: 392 212.71.0.60:5070 "Noisy feedback tells: pid=18743 req_src_ip=213.219.137.137 req_src_port=5070 in_uri=sip:ser.edpnet.net:5070 out_uri=sip:ser.edpnet.net:5070 via_cnt==1". . ----------------------------------------------- 5060 ----------------------------------------------- filter: ip and ( port 5060 ) # U 213.219.137.137:5060 -> 212.71.0.60:5060 REGISTER sip:ser.edpnet.net:5060 SIP/2.0. Via: SIP/2.0/UDP 213.219.137.137:47726. Supported: replaces. User-Agent: SIP201 (lp201sip.100a). Contact: <sip:bart@10.0.0.2:5060>;expires=60. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13c4-0-429-495. To: <sip:bart@ser.edpnet.net>. Call-ID: a000002-13c4-0-401-719e-1. CSeq: 1 REGISTER. Content-Length:0. . # U 212.71.0.60:5060 -> 213.219.137.137:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 213.219.137.137:47726;rport=5060. From: <sip:bart@ser.edpnet.net> ;tag=a000002-13c4-0-429-495. To: <sip:bart@ser.edpnet.net>;tag=61a88e7fd5f0561d96cde0cc9ecba6d7.2508. Call-ID: a000002-13c4-0-401-719e-1. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="ser.edpnet.net", nonce="40fcccbe3b4e06bc429de0a886d7b43409cb8427". Server: Sip EXpress router (0.8.13-dev-33-usrloc (i386/linux)). Content-Length: 0. Warning: 392 212.71.0.60:5060 "Noisy feedback tells: pid=18727 req_src_ip=213.219.137.137 req_src_port=5060 in_uri=sip:ser.edpnet.net:5060 out_uri=sip:ser.edpnet.net:5060 via_cnt==1". _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] developer related question
by Richard 20 Jul '04

20 Jul '04

Ser has a module cpl-c which supports cpl. There is a rfc for cpl, just google it. It's very easy to read. --- Joseph Cheung <Joseph.Cheung2(a)TELUS.COM> wrote: > Question now is how to get the support for cpl? > > Can you point me to the right uri? > > Thanks > -----Original Message----- > From: Richard [mailto:mypop3mail@yahoo.com] > Sent: July 19, 2004 9:51 PM > To: Joseph Cheung; serusers(a)lists.iptel.org > Subject: Re: [Serusers] developer related question > > You can use a cpl to each subscriber. Then it can > forward, redirect or reject the request according to > the cpl script, which is a much flexible solution. > > Richard > > > --- Joseph Cheung <Joseph.Cheung2(a)TELUS.COM> wrote: > > Hi. > > > > I think there is another email address I would use > > for this kinds of question. If so, much appreciate > > if someone could let me know. > > > > Here is my question. > > > > I have created a module that will lookup the > > database and determine if I should forward, > > redirect, or reject the request. > > > > Since there is no switch statement in the script, > I > > am wondering if I would save the "action" into a > > static variable and export a function to check and > > determine the action. > > > > I know the above could be done and it should be ok > > as we do a fork - should duplicate the memory as > > well. > > > > I would much appreciate if someone could give me a > > YES or NO answer or any idea how this is done. > > > > Many thanks. > > > > -----Original Message----- > > From: serusers-bounces(a)lists.iptel.org > > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of > > Jiri Kuthan > > Sent: June 19, 2004 2:01 PM > > To: Steve Blair; serusers(a)lists.iptel.org > > Subject: Re: [Serusers] CC-Diversion > > > > This may help: > > > http://www.iptel.org/ser/doc/seruser/seruser.html#AEN1018 > > > > -jiri > > > > At 04:09 PM 6/17/2004, Steve Blair wrote: > > > > > Hello: > > > > > > Can anyone speak to how the CC-Diversion > field > > is used > > >by Cisco gateways? It is my understanding that > > adding this header > > >field in SER will result in the Cisco gateway > > setting the calling party > > >id to the value in the CC-Diversion header. Is > this > > correct? > > > > > > I'm asking because we have an Octel 350 voice > > mail system that > > >I would like to use for mailbox for IP phone > users. > > The 350 expects > > >a physical SMDI circuit for the call signaling > path > > and a different > > >circuit for the message body path. It would be > > great to allow the > > >SIP signaling exiting our IP cloud through a > Cisco > > gateway to be > > >able to tell the Octel system to which subscriber > > mailbox the message > > >should be delivered. > > > > > >Thanks,Steve > > > > > >-- > > > > > >ISC Network Engineering > > >The University of Pennsylvania > > >3401 Walnut Street, Suite 221A > > >Philadelphia, PA 19104 > > > > > > > > >voice: 215-573-8396 > > > 215-746-7903 > > > > > >fax: 215-898-9348 > > > > > >sip:blairs@upenn.edu > > > > > >_______________________________________________ > > >Serusers mailing list > > >serusers(a)lists.iptel.org > > >http://lists.iptel.org/mailman/listinfo/serusers > > > > -- > > Jiri Kuthan http://iptel.org/~jiri/ > > > > _______________________________________________ > > 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 > > > > > > > __________________________________ > Do you Yahoo!? > Vote for the stars of Yahoo!'s next ad campaign! > http://advision.webevents.yahoo.com/yahoo/votelifeengine/ > > > > __________________________________ Do you Yahoo!? Vote for the stars of Yahoo!'s next ad campaign! http://advision.webevents.yahoo.com/yahoo/votelifeengine/

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Re: [Serusers] record route question
by Samuel Osorio Calvo 20 Jul '04

20 Jul '04

Only SIP messages. Signalling and RTP have completely independent paths. Unclassified >>> "Lakmal Silva" <lakmal(a)lankacom.net> 07/20/04 06:27AM >>> Hi all, I have a question on record routing. Once record routing is done, will the whole traffic (including rtp) will be routed through SER or only all the SIP messages will be passed through SER? -- Regards, Lakmal Lankacom Services (Pvt) Ltd. 65C, Dharmapala Mawatha, Colombo 07. Sri Lanka. Tel: +94-11-2437545 www.lankacom.net _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] play a message before forwarding
by Leon de Rooij 20 Jul '04

20 Jul '04

Hi, I've got ser working great over here, but I cannot find out how I can play a message before forwarding the call. Like this: - Person A calls person B via the SER proxy - SER proxy forwards to mediaserver - Mediaserver plays a message (like 'this call is free of cost') - Mediaserver hangs up - SER proxy forwards the call to person B - Person A and person B can talk to each other now.. Is this possible ? Thanks, Leon de Rooij leon(a)scarlet-internet.nl

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Re: [Serusers] record route question
by Lakmal Silva 20 Jul '04

20 Jul '04

Thanks you all of you who replied so promptly. > Only SIP messages. Signalling and RTP have completely independent > paths. > > Unclassified >>>> "Lakmal Silva" <lakmal(a)lankacom.net> 07/20/04 06:27AM >>> > Hi all, > > I have a question on record routing. Once record routing is done, will > the > whole traffic (including rtp) will be routed through SER or only all > the > SIP messages will be passed through SER? > > > -- > Regards, > Lakmal > > Lankacom Services (Pvt) Ltd. > 65C, Dharmapala Mawatha, > Colombo 07. > Sri Lanka. > Tel: +94-11-2437545 > www.lankacom.net > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > -- Regards, Lakmal Lankacom Services (Pvt) Ltd. 65C, Dharmapala Mawatha, Colombo 07. Sri Lanka. Tel: +94-11-2437545 www.lankacom.net

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sr-users July 2004