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[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 30 Oct '06

30 Oct '06

Please ignore previous post which I sent before completing! I have two problems: 1. A 2nd Remote-Party-ID header gets appended when I try to do two subst() on it. 2. Xlog seems to only have a ref to the orignal unedited rpid header. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite incoming calling numbers from my PBX to be in 5-digit format and other US 10-digit numbers into proper E.164 format since my PBX and at least one of my ITSPs doesn't provide them as E.164. So, I do a couple of subst() operations on INVITEs. What's wierd is that one Remote-Party-ID header becomes two when it gets to the UA. The call flow is: INVITE r-uri <sip:10508@128.59.59.96:5060> from <sip:9174147124@128.59.59.242> With the original RPID header as: Remote-Party-ID: <sip:9174147124@128.59.59.242> And after route[10] the headers that end up on the UA are: Remote-Party-ID: <sip:9174147124@siptest.columbia.edu> Remote-Party-ID: <sip:+19174147124@128.59.59.242> It looks like the 2nd subst() happened against the original RPID, not the edited one and somehow two RPID headers got created. Here's the code for the route block: route[10] { xlog("L_INFO","route[10] $ci: method $rm r-uri <$ru> from <$fu> rpid <$re>\n"); # first just clean out the domain name subst('/^(Remote-Party-ID:.*<sip:.*)@.*>(.*)$/\1(a)siptest.columbia.edu>\2/'); # now strip our numbers down to 5 digits subst('/^(Remote-Party-ID:.*<sip:)(\+)?(1)?21285([134][0-9]{4})@/\1\2/'); # expand 10-digit NANP numbers to E.164 subst('/^(Remote-Party-ID:.*<sip:)([0-9]{10}.*$)/\1+1\2/'); # ??? xlog doesn't show the result of rewriting it. # xlog("L_INFO","route[10] RPID is now <$re>\n"); xlog("L_INFO","route[10] $ci: RPID was rewritten.\n"); return; } #end of route[10] Any help with this would be greatly appreciated! /a

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[Users] openser.cfg
by simo med 30 Oct '06

30 Oct '06

dear sir or madame; i'm trying to start an openser, but i can't, i need to configure properly my openser.cfg! if you have an example of how can i configure my openser.cfg, to make it work, i will very thanksfull simo

2 1

[Users] How to xlog failure status in failure_route?
by Alan Crosswell 30 Oct '06

30 Oct '06

I tried xlog $rs $rr to log the reason for a failure in failure_route but it just logs NULL. What am I doing wrong? The error status is available to test with t_check_status and is logged by ACC but not xlog. # Error from media gateway failure_route[4] { xlog("L_INFO","failure_route[4] $si: method $rm r-uri <$ru>\n"); if (t_check_status("(5..)|(6..)|(403)")) { # gateway failure/refusal xlog("L_INFO","failure_route[4] gateway returned 5xx or 6xx. Status: $rs $rr\n"); if (!next_gw()) { xlog("L_CRIT","No more gateways for <$tu>\n"); t_reply("503", "Service not available - No more gateways"); return; } else { xlog("L_INFO","failure_route[4] next gw r-uri <$ru> \n"); t_on_failure("4"); t_relay(); return; } } xlog("L_INFO","failure_route[4]: relaying normal error\n"); t_relay(); } Oct 26 18:02:04 jello /usr/sbin/openser[2028]: ACC: call missed: method=INVITE, i-uri=sip:9319174147124@siptest.columbia.edu;user=phone, o-uri=sip:19174147124@67.151.91.22:50 60;transport=udp, call_id=c141e612-75cfa484-cfce783b(a)128.59.37.218, from="Alan Crosswell" <sip:10508@siptest.columbia.edu>;tag=CA5BDB76-6DB44EED, code=403 Forbidden Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] 128.59.37.218: method INVITE r-uri <sip:19174147124@67.151.91.22:5060;transport=udp> Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] gateway returned 5xx or 6xx. Status: <null> <null> Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] next gw r-uri <sip:19174147124@67.151.91.22:5060;transport=udp>

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[Users] Adapting the openserctrl file to a UNIX odbc client...
by Marvin Ottley 30 Oct '06

30 Oct '06

Hello all, I am looking for some information/pointers on how to adapt the openserctl file to use a UNIX ODBC client. Thanks, Marvin

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[Users] Ring group
by Marcos Biscaysaqu 29 Oct '06

29 Oct '06

Hi there. I just getting my Opserser server up and I want to be able to create child accounts from my system to be able to ring multiple extension at the same time, is this possible ? Thanks Marcos Biscayaqu

2 1

[Users] openser and PSTN
by geovani ferronetti 29 Oct '06

29 Oct '06

Hi, I want to connect OpenSER with PSTN, as media gateway I use Cisco AS5300, vfc c542. I am interested about config part from OpenSER. Where I can get some documentation, information? Thx __________________________________________________ Do You Yahoo!? Poco spazio e tanto spam? Yahoo! Mail ti protegge dallo spam e ti da tanto spazio gratuito per i tuoi file e i messaggi http://mail.yahoo.it

2 2

[Users] yate client windows
by sinclair fils 29 Oct '06

29 Oct '06

hello, all someone can tell me how i can configurate yate client to work which openser when i make a config the account cant register please help ___________________________________________________________________________ Découvrez une nouvelle façon d'obtenir des réponses à toutes vos questions ! Profitez des connaissances, des opinions et des expériences des internautes sur Yahoo! Questions/Réponses http://fr.answers.yahoo.com

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[Users] Why is a second RPID header created when I do two subst()?
by Alan Crosswell 28 Oct '06

28 Oct '06

I have two problems: 1. I am attempting to clean up my Remote-Party-ID header so that my Polycom UAs will display the incoming number as a number rather than as a URI. This requires the domain to match that of the UA's server (e.g. siptest.columbia.edu). And, I want to rewrite incoming calling numbers from my PBX to be in 5-digit format and other US 10-digit numbers into proper E.164 format since my PBX and at least one of my ITSPs doesn't provide them as E.164. So, I do a couple of subst() operations on INVITEs. What's wierd is that one Remote-Party-ID header becomes two when it gets to the UA. The call flow is: INVITE r-uri <sip:10508@128.59.59.96:5060> from <sip:9174147124@128.59.59.242> rpid <sip:9174147124@128.59.59.242> Here's the code for the route block: route[10] { xlog("L_INFO","route[10] $ci: method $rm r-uri <$ru> from <$fu> rpid <$re>\n"); # first just clean out the domain name subst('/^(Remote-Party-ID:.*<sip:.*)@.*>(.*)$/\1(a)siptest.columbia.edu>\2/'); # now strip our numbers down to 5 digits subst('/^(Remote-Party-ID:.*<sip:)(\+)?(1)?21285([134][0-9]{4})@/\1\2/'); # expand 10-digit NANP numbers to E.164 subst('/^(Remote-Party-ID:.*<sip:)([0-9]{10}.*$)/\1+1\2/'); # XXX dip the calling name database XXX # ??? xlog doesn't show the result of rewriting it. # xlog("L_INFO","route[10] RPID is now <$re>\n"); xlog("L_INFO","route[10] $ci: RPID was rewritten.\n"); return; } #end of route[10]

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[Serusers] Accounting problems after mysql backup restore
by C. Mewes 28 Oct '06

28 Oct '06

After backup and restoring MySQL the accounting did not worked anymore. I could make calls but the amount does not show up. The problem is that o_uri has the format sip:00 + codigo de tarifa pero en la base de datos se esta guardando sin el 00 solamente sip:+codigo de tarifa ( la razon no la se, no he modificado nada de eso ). After changig in accounting $codigo2 = "sip:".$row2->codigo; if ( strncmp ( $codigo2, $row->o_uri, strlen($codigo2) ) == 0) { $tarifa=$row2->tarifa; $nombre=$row2->nombre It works again.What is wrong??? Thanks to everybody who can help. CFM

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[Users] failure route doubts
by Jose Silva 27 Oct '06

27 Oct '06

Hi, I'm having some doubts on how to use/do some things with dispatcher module and failure route. I'll try to explain my problem. On my main route block I select the destination (using ds_select_dst) to where I want to send the message, and indicate that responses should pass through the failure_route, I also add a header with some things that are needed by the next proxy. On the failure route when I get 408 Time Out I mark the previous destination and select the next destination. until here everything goes fine. After this I want to change the message that I'm going to send to next proxy, for this I need to remove the header that I inserted previously on the main route block and access the destination URI that the ds_next_dst updates. My first doubt is that when I print the destination URI it has the previous destination ( the one used in the main route block) but when it relay's the message it goes to the right destination (the one returned from ds_next_dst), Is there any reason for this ? I'm using $du to access it, is this wrong? My second doubt is, what sip message do I have access to in the failure route (the one I received or the one I sent out)? I ask this because I try to remove the header I inserted in the main route block and it does not remove anything but when I relay the message the header I tried to remove goes on the message. Thanks in advance, José Carlos Silva

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