sr-users January 2005

sr-users@lists.kamailio.org

199 participants
399 discussions

Re: [Serusers] Support for Gateway still on RFC2543

by Klaus Darilion

Simon Miles wrote: > Klaus, > > Thanks for the feedback, but I still think it is a problem. > > If I use the prefix command, this effects the URI field but not the To > field. According to RFC2543 the To field is the one to use for dialling > when the INVITE gets to it's final destination. Are you sure that the To: field is used fpr dialing - not the request URI? Can you point me to the relevant sections in RFC2543? regards, klaus PS: Please CC to the list. > > Hence the prefix command can't be used ! ! ! If I mangle the To field > then this effects the Call-ID so the SIP software sees a reply to the > INVITE as another message ! ! > > Simon > > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: 10 January 2005 22:27 > To: Simon Miles > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] Support for Gateway still on RFC2543 > > > There should be no problem at all - RFC 3261 is compatible with the old > RFC. ser will look for the "lr" parameter in the via headers and will > use strict routing if the lr parameter is not found in the topmost via > header. > > regards, > klaus > > Simon Miles wrote: > > >>Dear Community, >> >>I still have gateways that confirm to RFC2543 and not the newer >>RFC3261. This means the use of URI and To fields are different. Is >>there any way of telling sip_router that it needs to conform to the >>old spec ? >> >>Thanks >> >> >>Simon >> >> >> >>---------------------------------------------------------------------- >>-- >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

20 years

[Serusers] IPv6, problems with link local

by Michelle Mendel

20 years

[Serusers] Regisater request with Recorde Route Header........

by innovation.interops＠wipro.com

Hello, As per RFC 3261 the Registrars must ignore Record-Route field in any incoming REGISTER request and should not include the same in the response. But I find SER implementations including the Record Route Field in the 200OK Response. Any advice or suggestion?? Thanks karthikeyan Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

20 years

[Serusers] Processing REGISTER REQ section 10.3

by innovation.interops＠wipro.com

Hello, As per section 10.3 step 6 the bindings must not be removed if there is an incoming REGISTER request with... CallID==One stored CSeq==Same as stored Contact : * Expires: 0 But I found SER Registrar implementations removing the bindings and sends 200Ok.It should have dropped the request. Any thoughts or inputs?? Thanks karthikeyan Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

20 years

[Serusers] AVP ops module question.

by Ricardo Martinez

Hello. I have a few questions about the usage of the AVP ops module for Dynamic Routes. First. Suppose that i have a table called "Prefix". This table looks like : uuid code Value 107 0056 192.168.0.10 107 0044 192.168.0.20 107 001 192.168.0.30 and maybe a few more lines like these. My first question is Does the avp_db_load( "i:107" ) command load all the "code" and "value" rows? Second : if i succesfully loaded all the rows i need to compare the RURI from the INVITE with the avp parameter. In the avp_check("i:107", "eq/$ruri/i ") i need to compare the ruri parameter with the loaded code. So if i loaded all the codes, the avp_check compare the ruri with all the avp's loaded?. Does the "eq/" check a exact match or can i use it for the "best" match?. For example 005621234567 is matched with 0056. I really hope that some can help me here. Thanks in advance Ricardo Martinez Ogalde.-

20 years

[Serusers] SER Radius Web FrontEnd

by Michele "O-Zone" Pinassi

Hi all, i'm start working to a Web (PHP+MySQL) frontend to manage users, view calls and accounting, in SER with Radius Auth and Acc. Before loosing time, there's something already done ? Thanks ! Oz -- ---- O-Zone ! No(C) since 1996 www.zerozone.it

20 years

[Serusers] routing question - what is "current URI" in these cases?

by Gavin Bensom

Hi, I'd like some clarification on script routing. To quote the SER Admin guide. . . " It is important to realize that ser operates over current URI all the time. If an original URI is rewritten by a new one, the original will will be forgotten and the new one will be used in any further processing. In particular, the uri matching operand and the user location action lookup always take current URI as input, regardless what the original URI was. " Question 1 - Current URI = sip:user@domain.com. What is the "current URI" after these two lines are processed in the script? What is the other uri in the destination set? rewritehostport("10.10.10.40:5095"); append_branch(); Question 2 - Current URI = sip:user@domain.com. What is the "current URI" after the same two lines are processed in the script but in the opposite order? What is the other uri in the destination set? append_branch(); rewritehostport("10.10.10.40:5095"); Question 3 - Current URI = sip:user@domain.com. What is the "current URI" after these three lines are processed in the script? Assume that user(a)domain.com has two locations. sip:user@10.10.10.20 - priority = 0.0 sip:<user_cell_phone>@<gateway_ip> - priority = 1.0 rewritehostport("10.10.10.40:5095"); append_branch(); lookup("location"); What are the other uri's in the destination set in this case? I'm confused and I've tried to determine the behavior by testing with different script configurations. The reason I'm asking is that I want my users to be able to put their cell phone numbers in with the form of sip:<cell_number>@mydomain.com Currently this does not work. To achieve proper cell phone forwarding the contact location must be: sip:<cell_number>@gateway_ip_address In previous script configurations the sip:cell_number@mydomain.com format worked. When I added these lines: rewritehostport("10.10.10.40:5095"); append_branch(); to my script before calling lookup("location"); Then SER started forwarding sip:cell_number@mydomain.com to my DNS server. :) Not what I wanted, obviously. My DNS server does have SRV records for SIP, but I'm not sure if I need to add a line in ser.cfg to make ser use SRV records. Any advice or enlightenment is appreciated. Thanks, G. __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

20 years

[Serusers] click-to-dial problems

by Gavin Bensom

Hi, I'm having a very hard time getting the serweb click to dial functionality working. When I log into serweb and click on a sip address in the phonebook page, the ctd sub-window pops up and does not display any text. After about 20-30 seconds, the window displays "408 Request Timeout" in red text. This is all that happens. I cannot see any sip INVITE messages flowing via ngrep on the server. I get a entry in /var/log/httpd/access_log that shows that ctd.php was called properly. See line from access_log below. 10.10.30.113 - - [18/Jan/2005:17:53:23 -0800] "GET /user/ctd.php?target=sip:rk@mydomain.com&uri=sip:gb@mydomain.com&kvrk=1106 099602000 HTTP/1.1" 200 1323 "http://voice/user/phonebook.php?kvrk=41eda110e9af5" "Mozilla/5.0 (Windows; U; Windows NT 5.1; en-US; rv:1.7.5 ) Gecko/20041107 Firefox/1.0" I've even tried adding an echo line into the php code in the "write2fifo" function. The echoed text shows up in the ctd window when the "408 Request Timeout" message appears. I've tried the ctd.sh shell script as well. It fails to complete and I must use ctrl-c to escape it and then kill it manually. My php variable[ $this->fifo_server="/tmp/ser_fifo"; ] matches my ser.cfg script. # script excerpt check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=5 fifo="/tmp/ser_fifo" #sip_warning=no group="serfifo" fifo_mode=0666 Any ideas? I can't find any other log that shows any activity when I attempt to use click-to-dial. I really want to get this working. Thanks, G. __________________________________ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com

20 years

[Serusers] More progress on my media proxy issue

by Doug Eubanks

Ok, I have gotten my media proxy to work fine between sip to sip calls, but I seem to have problems when I am forwarding a call to the PSTN. I am sure I just have something in the wrong order, any pointers? Earlier in the ser.cfg if (method=="INVITE" || method=="ACK") { log(1, "[SER]: Loose Route detected, using MediaProxy\n$ use_media_proxy(); }; Forwarding section..... rewritehostport("PSTNGATEWAY:5060"); forward(uri:host, uri:port); break; The speed of time is one-second per second.

20 years

RE: [Serusers] multiple SIPURA behind same NAT

by Richard

What firewall do you have? Your firewall probably has problems to support hairpin traffic. Richard _____ From: Raymond Chen [mailto:rchen@cityabove.net] Sent: Tuesday, January 18, 2005 4:56 PM To: 'Richard' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT We use SIPURA 2000 and rtpproxy. I have attached the sip message log. Ray _____ From: Richard [mailto:richard@o-matrix.org] Sent: Wednesday, January 19, 2005 10:27 AM To: 'Raymond Chen' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT What nat device do you have? Does it have any sip helper module? Are you running rtpproxy? It will be helpful if you can send a sip packet dump. Richard _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Raymond Chen Sent: Tuesday, January 18, 2005 2:59 PM To: 'Matt Schulte' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT Matt, Both port use different sip port, and can make calls at the same time. But still can't get voice when call to each other. Raymond _____ From: Matt Schulte [mailto:mschulte@netlogic.net] Sent: Tuesday, January 18, 2005 9:08 AM To: Raymond Chen; serusers(a)lists.iptel.org Subject: RE: [Serusers] multiple SIPURA behind same NAT If I had a guess I would say either the NAT or rtp/media proxy is getting confused about which client is which. Ignoring the problem for a second, can both sipura's make calls at the same time? Are you using NAThelper for this with SER? You mentioned "without proxy", this would mean port forwarding? Try this, make each sipura use a diff range of RTP ports (in their config). Matt -----Original Message----- From: Raymond Chen [mailto:rchen@cityabove.net] Sent: Monday, January 17, 2005 12:25 PM To: serusers(a)lists.iptel.org Subject: [Serusers] multiple SIPURA behind same NAT Dear all, I have two sipura box behind NAT. both use via header and can dial out and received calls without proxy. but they have no voice when they call to each other, any idea? Raymond

20 years

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sr-users January 2005