sr-users December 2007

sr-users@lists.kamailio.org

130 participants
234 discussions

by CSB

For some reason I cannot get allow_trusted() to evaluate to true for a matching URI. Relevant sections of the config (OpenSER 1.1): modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", "mysql://username:password@localhost/ser") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") route { if (!allow_trusted()) { xlog("!allow_trusted. rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); }; } The trusted table contains the following: +----------------+-------+--------------+------+ | src_ip | proto | from_pattern | tag | +----------------+-------+--------------+------+ | 202.200.20.200 | any | NULL | NULL | +----------------+-------+--------------+------+ 1 row in set (0.00 sec) I can see that the following query is being executed: select proto,from_pattern,tag from trusted where src_ip='202.200.20.200' But no match is made because the log file shows: Dec 17 15:37:56 beta /usr/local/sbin/openser[7209]: !allow_trusted. rs <null> si 147.200.20.200 rm INVITE ru sip:5551234@147.200.20.200 tu sip:5551234@147.200.20.200 fu sip:021446202@202.200.20.200 rr <null> Could anyone provide any suggestions on what I'm doing wrong? Regards Cameron

16 years, 11 months

[OpenSER-Users] [Fwd: IMC Module]

by Amit Vijayvargiya

16 years, 11 months

[OpenSER-Users] OpenSER billing options

by Andy Smith

Hi list, are there alternatives to Radius for billing integration with OpenSER? The problem Radius seems to pose me is that it is per user and therefore requires to have a full database of every softphone/SNOM/etc user, which is admin intensive. Does anyone know if there is a method of accounting data per SIP peer (for example per Asterisk system attached to OpenSER) which could record call details without needing to know about all the actual handset users? I suppose you can do anything you want with the accounting module so this is really a "does anyone already do this" question so I dont need to invent everything from scratch... thanks for any suggestions, Andy.

16 years, 11 months

[OpenSER-Users] Problem Registering via TCP

by Krom

I have a UA that I am attempting to register to OpenSER 1.1.1 via TCP and the only hint I can give it the following statement that is placed in the syslog file. WARNING: io_wait_loop_sigio_rt: ignoring event 41 on fd 0 (fm->fd=0, fm->data=(nil)) I have placed in the first line of the main route the following just to see if the registration is making it to the main route and it is not. if(proto==TCP) { xlog("L_INFO","SIP message was received over TCP\n"); }; I can tell from tcpdump that the packet is being received and a complete SYN,ACK is successful. Global Config: listen=10.1.1.28:5060 disable_tls=1 tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 A netsat shows the port in a listening state. When starting OpenSER it shows that it is listening on TCP:IP_ADDR:5060 When the UA sends a register request, a connection is established, the one statement above is placed in the log file and then the connection goes to a CLOSE_WAIT state. Any suggestions. I know at least one of us has seen this before. I searched the list archives and found one thread in Nov 06 discussing the same issue; but no follow-up of resolution posted. http://lists.openser.org/pipermail/users/2006-November/007653.html -- -- Krom kyle.romulas(a)gmail.com

16 years, 11 months

[OpenSER-Users] CPL doesn't allow "redirect" to url="http://webpage"

by Iñaki Baz Castillo

Hi, I want to set a redirect to a web page in CPL but I get error: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> <notfound> <location url="http://google.es"> <redirect /> </location> </notfound> </lookup> </incoming> </cpl> ~# openserctl fifo LOAD_CPL sip:user@domain /etc/openser/cpl/user(a)domain.xml 500 Bad CPL file Log:: Error: Encoding of the CPL script failed It works if I set (as usual): <location url="sip:user2@domain"> Why doesn't CPL module allow redirection to non SIP target? -- Iñaki Baz Castillo

16 years, 11 months

[OpenSER-Users] OpenSER with multidomain support

by satish patel

Dear all I have post many question about multidomain support help for openser i need cfg file which you ppl are useing caz openser.cfg is bit harder for me and i have also find many help on ser on google but still i m confuse about parmeter and options if anybody have done this configuration send it me or explain me how to do it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

16 years, 11 months

[OpenSER-Users] get Count of items in lists

by Dave Singer

using openser 1.3, is there a way to get a count of how many items there are in an avp variable other than: while ( is_avp_set("$avp(s:fork_uri)") ) { $var(fork_uri_count) = $var(fork_uri_count) + 1 ; avp_copy("$avp(s:fork_uri)", "$avp(s:fork_uri_rev_order)/d") ; } then if you wanted things in the same order as they were originally you would also have to: while ( is_avp_set("$avp(s:fork_uri_rev_order)") ) { avp_copy("$avp(s:fork_uri_rev_order)", "$avp(s:fork_uri)/d") ; } talk about an expensive counting Several ways I could see this implimented: New function avp_count("$avp(s:myvar)", "$avp(s:count_result)") or avp_count("$avp(s:myvar)") where $rc contains the result. is_avp_set("$avp(s:myvar)") where $rc is set to the count, probably best as you could test if exitst and the go right to processing. avp_check("$avp(s:myvar)", "cn/g") where $rc is set to the count. It could always set $rc to count of found matches for when the g flag is used. Of course using $rc for is_avp_set() or avp_check() depends on if setting that to non-zero makes a test like if( is_avp_set("$avp(s:myvar)") ) always trigger as false. also on getting a count ... another place I need to get a count is how many times a character(s) (or shoot the moon, a regex) occurs in a string. for handeling things like transformations like {s.select,index,seperator}, eg: $(var(findme){ s.select,$var(count),,}) sidenote ... (not sure what values for seperator are valid) a transformation like {s.count,seperator} Here is an example of something I was trying to do that would have benifited from knowing the count of seperator. I think that I got a warning or error when the test for the while indexed past the end of the string. xlog("L_INFO", "Original findme list to parse: $avp(s:findme)\n"); if ( $avp(s:findme) =~ "\|" ) { $var(findme) = $(avp(s:findme){s.select,0,|}) ; $var(findme_tm_out) = $(avp(s:findme){s.select,1,|}) ; xlog("L_INFO", "This findme has a timeout: $var(findme_tm_out)\n"); } else { $var(findme) = $avp(s:findme) ; } xlog("L_INFO", "findme list to parse: $var(findme)\n"); $var(findme_cnt) = 0 ; while ($(var(findme){s.select,$var(findme_cnt),,}{s.len }) > 0 ) { xlog("L_INFO", "Loop $var(findme_cnt), Adding $(var(findme){s.select,$var(findme_cnt),,}) to TO_list\n"); $avp(s:TO_list) = $(var(findme){s.select,$var(findme_cnt),,}) ; $var(findme_cnt) = $var(findme_cnt) + 1 ; } can/do $var()'s stack like $avp()'s? Thanks Dave

16 years, 11 months

Re: [OpenSER-Users] Missed calls (I?aki Baz Castillo)

by David Villasmil Govea

te refieres a ésto? ya está en el openser.cfg if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; > > > El Friday 14 December 2007 13:55:01 David Villasmil Govea escribi?: > > * Furthermore,?Openser doesn't see all INVITES+BYE, so i can't make > cdrs!! > > I've read somewhere that to log all transaction messages i should have > > "strict route" as oppossed to "loose route". > > No, you need to add "Record-Route" header. This is done by > "record_route()" > function of rr module. > > -- DVG -- Imagination is more important than knowledge Albert Einstein

16 years, 11 months

Re: [OpenSER-Users] Users Digest, Vol 31, Issue 46

by David Villasmil Govea

16 years, 11 months

[OpenSER-Users] openser config issue or asterisk bug

by Patrick Baker

Internal UA's connecting to the sip proxy work fine, but as soon an External UA connects to the sip proxy via a NAT'd public IP address - the call lasts for approx 20 seconds then disconnects. From what I can see openser is attempting to respond to asterisk after the session is initiated via the public IP NAT'd address - and is unable since asterisk and openser are behind the firewall on the internal network ( ELBOW routing ). Any help getting this working would be most appreciated, I would also like to document the issue to help others that encounter similar problems as mine. External phones: UA -> [(public IP) Firewall (NAT to localip)] -> openser -> asterisk Internal phones: UA -> openser -> asterisk #### Layout ( asterisk and openser are on the same machine ) NATD public ip = xxx.206.xxx.136 -> 10.3.1.33 phone ip = xxx.206.xxx.137 openser = 10.3.1.33 p5060 asteriskSIP = 10.3.1.34 p5060 #### Asterisk SIP Config [general] canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5060 bindaddr=10.3.1.34 sipdebug=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all host=10.3.1.33 #### SER LOG Dec 13 22:15:03 phonesys-slave openser[15976]: New request - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.34:5060;transport=udp F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 #### Asterisk LOG *CLI> [Dec 13 22:15:03] <--- SIP read from 10.3.1.33:5060 ---> INVITE sip:500@10.3.1.34:5060;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> CSeq: 2 INVITE Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Contact: <sip:pbaker@xxx.206.xxx.137> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 254 v=0 o=- 978310717 978310717 IN IP4 xxx.206.xxx.137 s=Polycom IP Phone c=IN IP4 xxx.206.xxx.137 t=0 0 m=audio 35022 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Dec 13 22:15:03] --- (16 headers 11 lines) --- [Dec 13 22:15:03] == Using TOS bits 0 [Dec 13 22:15:03] == Using CoS mark 5 [Dec 13 22:15:03] Sending to 10.3.1.33 : 5060 (no NAT) [Dec 13 22:15:03] Using INVITE request as basis request - f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 [Dec 13 22:15:03] No user 'pbaker' in SIP users list [Dec 13 22:15:03] Found peer 'openser' for 'pbaker' from 10.3.1.33:5060 [Dec 13 22:15:03] Found RTP audio format 0 [Dec 13 22:15:03] Found RTP audio format 8 [Dec 13 22:15:03] Found RTP audio format 18 [Dec 13 22:15:03] Found RTP audio format 101 [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Found audio description format PCMU for ID 0 [Dec 13 22:15:03] Found audio description format PCMA for ID 8 [Dec 13 22:15:03] Found audio description format G729 for ID 18 [Dec 13 22:15:03] Found audio description format telephone-event for ID 101 [Dec 13 22:15:03] Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Dec 13 22:15:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Looking for 500 in default (domain 10.3.1.34) [Dec 13 22:15:03] list_route: hop: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> [Dec 13 22:15:03] <--- Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Length: 0 <------------> [Dec 13 22:15:03] -- Executing [500@default:1] Playback("SIP/xxx.206.xxx.136-090b5380", "demo-abouttotry") in new stack [Dec 13 22:15:03] Audio is at 10.3.1.34 port 47068 [Dec 13 22:15:03] Adding codec 0x4 (ulaw) to SDP [Dec 13 22:15:03] Adding codec 0x8 (alaw) to SDP [Dec 13 22:15:03] Adding codec 0x100 (g729) to SDP [Dec 13 22:15:03] Adding non-codec 0x1 (telephone-event) to SDP [Dec 13 22:15:03] <--- Reliably Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 13 22:15:04] -- <SIP/xxx.206.xxx.136-090b5380> Playing 'demo-abouttotry.slin' (language 'en') [Dec 13 22:15:04] Retransmitting #1 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:05] Retransmitting #2 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:07] Retransmitting #3 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:11] Retransmitting #4 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:15] Retransmitting #5 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:18] -- Executing [500@default:2] Dial("SIP/xxx.206.xxx.136-090b5380", "IAX2/guest@misery.digium.com/s@default") in new stack [Dec 13 22:15:18] -- Called guest@misery.digium.com/s@default [Dec 13 22:15:18] -- Call accepted by 216.207.245.8 (format gsm) [Dec 13 22:15:18] -- Format for call is gsm [Dec 13 22:15:18] -- IAX2/216.207.245.8:4569-1 is ringing [Dec 13 22:15:19] -- IAX2/216.207.245.8:4569-1 answered SIP/xxx.206.xxx.136-090b5380 [Dec 13 22:15:19] Retransmitting #6 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2337 retrans_pkt: Maximum retries exceeded on transmission f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 for seqno 2 (Critical Response) [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up call f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 - no reply to our critical packet. [Dec 13 22:15:23] -- Hungup 'IAX2/216.207.245.8:4569-1' [Dec 13 22:15:23] == Spawn extension (default, 500, 2) exited non-zero on 'SIP/xxx.206.xxx.136-090b5380' [Dec 13 22:15:24] Really destroying SIP dialog 'f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137' Method: INVITE #### PCAP Diagram |Time | xxx.206.xxx.137 | 10.3.1.33 | |5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.058 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.058 | Status: 407 Proxy A |SIP: Status: 407 Proxy Authentication Required | |(5060) <------------------ (5060) | |5.140 | Request: ACK sip:50 |SIP: Request: ACK sip:500@xxx.206.xxx.136:5060 | |(5060) ------------------> (5060) | |5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.165 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |6.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |7.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |9.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |13.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |17.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |21.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | #### OpenSER Config 1.2 ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Thu Dec 13 22:43:43 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Thu Dec 13 22:43:43 +0100 2007' ######################################################################## listen = udp:10.3.1.33:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = yes check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); }

16 years, 11 months

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sr-users December 2007