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37518 discussions

Kamailio + rtpengine + Freeswitch on public/private network
by Eric Ravelo 20 Mar '21

20 Mar '21

Dear Kamailians, I have Kamailio+rtpengine as a dispatcher in front of FreeSWITCH. Kamailio and FreeSWITCH are on the same private network (in an AWS VPC). Kamailio receives calls on the Public Internet IP address and port, and dispatches to FreeSWITCH on the private network. All calls are coming from a SIP Trunk provider, from the Internet so no NAT on this side. My Hello World scenario is: FreeSWITCH answers the calls, plays a media file and disconnects. Problems: 1) In the INVITE relayed from Kamailio to FreeSWITCH, I can see a Record-Route with the public IP of Kamailio. Hence, the BYE sent by FreeSWITCH is sent to this public IP, rather than going back to the Kamailio path, and the call never gets completed from the caller's point of view. I tried adding record_route() in request_route{} but nothing changes. 2) The SDP part of all the messages coming back and forth from Kamailio and FreeSWITCH all contain the correct private IPs, but still the media file can't be heard on the caller's side. Is RTPengine correctly bridging the RTPs? Kamailio configuration is here: https://gist.github.com/eravelo/b3f026723e27d42354804ef942b34a04 Rtpengine configuration file has: [rtpengine] table = 0 interface = internal/10.20.1.158;external/10.20.1.158!KAMAILIO_PUBLIC_IP listen-ng = 127.0.0.1:2223 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 16384 port-max = 32768 Your help is appreciated. Thank you. Eric R.

2 2

received parameter in route URI not used despite path use_received set to 1
by Easy Rider 19 Mar '21

19 Mar '21

Hi I'd like to make sure I've understood correctly how use_received in path module is supposed to work, because I don't get what I expected to get after reading the documentation I've got a little set up with a UAC behind a NAT registering to a kamailio registrar through a kamailio edge proxy (he UAC is a simple sipp script simulating a UAC behind a NAT) path is enabled on both EP and registrar, and use_received is set to 1 on EP the path, contact and via headers in the REGISTER going from the EP to the registrar are : Via: SIP/2.0/UDP 192.168.26.129;branch=z9hG4bK20da.9f4ab6ad25eae6acdd4bd4d655b320ce.0 Via: SIP/2.0/UDP 10.0.0.5:6060 ;received=192.168.26.1;rport=23456;branch=z9hG4bK-105-1-1 Contact: <sip:1000@10.0.0.5:6060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS" Path: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> where 192.168.26.129 is obviously the edge proxy then I place a call to the UAC through registrar the EP receiveds the following INVITE : INVITE sip:1000@10.0.0.5:6060 SIP/2.0 Record-Route: <sip:192.168.26.142;lr> Via: SIP/2.0/UDP 192.168.26.142;branch=z9hG4bKdf25.142eebc72e940d07ad983a858c67f64e.0 Route: <sip:inside@192.168.26.129;lr;received=sip:192.168.26.1:23456> Via: SIP/2.0/UDP 192.168.26.1:34567 ;received=192.168.26.1;rport=34567;branch=z9hG4bK-107-1-1 From: "1001" <sip:1001@my-fake-domain.com>;tag=107-1 To: <sip:1000@my-fake-domain.com;user=phone> CSeq: 1 INVITE Call-ID: 1-107(a)192.168.26.1 Contact: <sip:1001@192.168.26.1:34567> ... The Route header is exactly what is expected, but the edge proxy still sends the request to the request-URI address (10.0.0.5:6060) instead of sending the request to the received address set in the first Route header The path module documentation states : 3.1. use_received (int) If set to 1, the “received” parameter of the first Route URI is evaluated and used as destination-URI if present. kamailio configuration on the EP is almost the default one (just added path support, and modified the REGISTRAR route to forward the request instead on saving the contact, so there is not even a mid registrar on the EP tried with kamailio 5.3.7 and 5.4.4 thanks

3 4

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 18 Mar '21

18 Mar '21

I have one Kamailio that makes an outbound register on two Asterisk. I have a dispatcher with that same two Asterisk destinations, on INVITE message I call replace_from("", ) in order to call a internal extension and on failure route I receive 401 and call the process of authorization is OK. The problem is that if the first Asterisk fail and I get 500 on failure route I can not call replace_from() because is not valid on the block. Is there any way to make this type of authorization process after receiving a 500 on failure route of dispatcher ? Below my cfg file. Thanks in advance. Antonio Couto #!KAMAILIO#!ifdef WITH_DEBUG#!define DBGLEVEL 3#!else#!define DBGLEVEL 2#!endif # - flags# FLT_ - per transaction (message) flags# FLB_ - per branch flags#!define FLT_ACC 1#!define FLT_ACCMISSED 2#!define FLT_ACCFAILED 3#!define FLT_NATS 5 #!define FLB_NATB 6#!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0log_prefix="{$mt $hdr(CSeq) $ci} " children=8 auto_aliases=noenable_tls=noenable_sctp=no tcp_accept_no_cl=yes # SIP port listen=udp:192.168.0.201:5060# xhttp port listen=tcp:127.0.0.1:8181 loadmodule "xhttp.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "htable.so" loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 1) # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params -----# default retransmission timeout: 2sec modparam("tm", "fr_timer", 1500)# default invite retransmission timeout after 1xx: 40sec modparam("tm", "fr_inv_timer", 40000) # ----- rr params -----# set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) # account triggers (flags) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:proxy@kamailio.org") modparam("dispatcher", "ds_ping_interval", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_timer_mode", 1) modparam("db_text", "db_mode", 1) modparam("db_text", "emptystring", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "track_cseq_updates", 1) modparam("uac", "restore_dlg", 1) modparam("uac", "reg_db_url", "text:///etc/kamailio/dbtext") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 60) modparam("uac", "reg_contact_addr", "192.168.0.201:5060") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations if (is_method("INVITE")) { route(DISPATCH); } else { sl_send_reply("501","Not Implemented"); exit; } } # Dispatch requests route[DISPATCH] { if ($si == "192.168.0.44" and $sp == 5060) { xlog("L_ALERT","from switch $si:$sp)\n"); $var(groupid) = "2"; } else{ xlog("L_ALERT","from others $si:$sp)\n"); $var(groupid) = "1"; } if(!ds_select_dst($var(groupid), "11")) { send_reply("503", "Service Unavailable"); exit; } if ($du =~ "192.168.0.210") { xlog("L_ALERT", "SCRIPT: uac_replace_from\n"); uac_replace_from("", "sip:4000@192.168.0.201:5060"); } xlog("L_ALERT", "SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { $var(ts) = t_get_status_code(); xlog("L_ALERT","ALERT: failure_route[RTF_DISPATCH]] from $si:$rp status:$var(ts)\n"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = "4000"; $avp(apass) = "1234"; $avp(arealm) = "asterisk"; if (uac_auth()) { route(RELAY); } exit; } if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if (ds_next_dst()) { xlog("L_ALERT", "retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) { return; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("500","Server Internal Error"); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # jsonrpc dispatch event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } }

2 2

[Question] About softphone for VoLTE
by shigeru ishida 18 Mar '21

18 Mar '21

Hi all, I have one question. What softphones can I use when trying VoLTE (not VoIP) with Kamailio? I would like to verify VoLTE of private LTE with Open5GS and Kamailio. I'm sorry to ask you a question without much understanding of Kamailio, VoIP and VoLTE. Thank you very much in advance. Best regards, --Shigeru

3 7

Server requirements estimation
by Marat Gareev 17 Mar '21

17 Mar '21

Hello! I'm configuring Kamailio proxy and trying to find instructions or advises for choosing the following options: *Host specific* - Number of cores - RAM size *Kamailio specific* - Number of children to fork for the UDP interfaces - Shared memory size The proxy will not connect to the database, only route INVITEs using the dispatcher. The expected load is 70K requests per hour. Are there any formulas for calculating the parameters?

2 2

sip_trace() in branch_route, kamailio 5.1
by Daniel Donoghue 17 Mar '21

17 Mar '21

Hello, Using kamailio 5.1, I am trying to use sip_trace(<internal-sip-uri>) from a branch route to send a copy of the current sip message to another location. Sending the message works fine but I note that the message being sent is the original incoming request and does not contain what was changed for the particular branch. Is there any way to get sip_trace() to send the content of the message for that specific branch? With every blessing, — Daniel Donoghue

2 1

Kamailio how to make UAC challenge on Dispatcher failure route
by Antonio Couto 17 Mar '21

17 Mar '21

2 1

kamcmd dlg.end_dlg not working in shell script
by Andrew Chen 16 Mar '21

16 Mar '21

Hi all, I have a script that cleans up hanging dialog and for some reason, executing 'kamcmd dlg.end_dlg' does not seem to work. Below I log events of stuck calls and as you can see it just keeps logging the same call until I manually run the kamcmd either in bash shell prompt or in kamcmd cli: 2021-03-08T19:00:04.347807+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:00:04 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:01:02.531822+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:01:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:02:02.689823+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:02:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:03:02.044960+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:03:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:04:02.436588+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:04:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:05:02.685572+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:05:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:06:02.775561+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:06:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:07:01.867021+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:07:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:08:01.888512+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:08:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:09:02.146887+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:09:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:10:02.649627+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:10:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:11:01.904948+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:11:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:12:01.932267+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:12:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:13:02.428128+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:13:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:14:02.617683+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:14:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:15:01.863099+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:15:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:16:01.650299+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:16:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:17:02.338576+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:17:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:18:01.743163+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:18:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:19:02.257369+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:19:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:20:01.624170+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:20:01 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul 2021-03-08T19:21:02.287343+00:00 sjointgkama51 /scripts/get-call-from-hepic.sh - get-call-from-hepic: Mon Mar 8 19:21:02 UTC 2021 Found active dialog on completed call. Delete call using h_entry: 1333 *h_id*: 11225 on SIP Call-ID 7j5r1msitcvg9siohoul Thoughts? Thanks! -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

5 15

dmq support for all transports - udp, tcp, tls, sctp
by Daniel-Constantin Mierla 16 Mar '21

16 Mar '21

Hello, (cross-posting - notification for developers, but also looking for testing support in the users community). So far dmq had support only for UDP, suitable to use for replication over trusted networks (local private network, or vpn). There were couple of discussions in the past about supporting other transport like tcp or tls. Earlier today I pushed several commits to dmq module trying to remove the limitation on UDP, and support the rest of the base protocols available in Kamailio (TCP, TLS and SCTP). Internally, a significant behaviour change is that the first (startup) peer notification is now done in the SIP worker one process, instead of the main process, in order to have all transports initialized. I could not spot any relevant side effect that can happen based on quick analysis, because the code with UDP only support was sending out the KDMQ requests, the corresponding replies could come back later. Anyhow, if any other dmq developer thinks of something not being quite right, reply to the sr-dev mailing list to discuss further. For users community: it would be appreciated if people using dmq can test with other transports (TLS, TCP, ...) and report any problems on sr-users mailing list or open an issue on bug tracker. Interesting test scenarios would be with replication of large amount of data or high rate of replicated requests per second. My testing got to the level of connecting to peers via TLS, trying to ensure the code is no longer limiting to UDP. I will do more, but I thought of giving a notification in early phase and get some feedback in order to avoid advancing too much in a wrong direction. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Dialog using Recod Route, with an asterisk
by Nicki Otte 15 Mar '21

15 Mar '21

Hi everyone I have a question regarding the dialog module. It uses the RecordRoute header, which seems all fine. The situation: I have a setup, where the Kamailio proxy, which is running with the dialog module, is talking to an Asterisk. Now the logic in the Asterisk, is to place a new call, and return in to the Kamailio proxy, which sends it to the PSTN. By the logic of the asterisk, it doesn’t persist with the RecordRoute headers. It starts from scratch with a new callId. Here lies the problem. Because the RecordRoute header isn’t given, then a new “dialog” is made, instead of using the one that is established. Here comes my question. Is it against the standards to make the Asterisk persist with these headers, or am I looking at the wrong solution? If not, I do see the power of the dialog module, when a call bounces between UAS’, but not when an asterisk is in the setup. Another solution could be to use a custom x- header for the dialog id, but it doesn’t look like that is a possibility. Have any of you run into this situation, and might have a solution? Thanks in advance // Nicki Bo Otte

2 2

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