Thanks very much Daniel, This is solving a part of my issue for now. Here is the routing section of my configuration file (it may be good to share it once complete as a minimally working proxy): route{ } if (!method=="REGISTER") record_route(); if (!loose_route()) { > $du = "sip:10.236.214.86:5060"; > #setdsturi("sip:10.236.214.86:5060"); #second way to do it > if (!t_relay()) { > sl_reply_error(); > } } > } I am trying to set up a kamailio proxy on an EC2 VM using this configurations and have been running a UAC and a UAS using SIPp on two other machines (UAC <--> Kamailio Proxy <--> UAS): UAC > sipp -sn uac -nr -r 1 -rp 1000 -d 0 -l 1 -p 5060 -trace_msg -i > SELF_IP(UAC) -rsa PROXY_IP:5060 UAS_IP:5060 UAS: It looks that messages go through, and are received by the server, but what I get back at the UAC (client) is a "dead call" error: Last Error: Dead call 1-1734(a)10.140.34.188 (aborted at index 8), receive... Is there a specific meaning to this "dead call" error? Is there anything that my proxy is missing in its routing or does it have to do with the UAC/UAS configs? Thanks Alireza On Tue, Jun 17, 2014 at 5:11 AM, Daniel-Constantin Mierla &lt;miconda(a)gmail.com > Hello, > listen is to specify local ip address or network interface on which > kamailio should listen for sip traffic. > To send to an ip address there are couple of variants, in config file: > $ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; > t_relay(); > exit; > Or, if you don't want to change the r-uri, then use: > $du = "sip:__NEXT_PROXY_IP__"; > t_relay(); > exit; > Of course, you have to replace the __NEXT_PROXY_IP__ with the appropriate > value. > More dynamic option would be using dispatcher module. > Cheers, > Daniel > On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote: > Dear Kamailio users, > I am trying to set up a simple scenario as follows: > UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC > I am new to Kamailio and had the following basic questions that came to > my mind after reading the documentation an the default kamilio.cfg config > script: > 1) Is there any good example of the above scenario in which the UAC and > UAS (caller and callee) are modeled using SIPp and proxies are simple > kamailio instances run on different machines that simply forward the SIP > packets? > 2) In specific, I want the proxies to simply forward the messages to > their downstream (I do not care about registration or other operations and > look for a minimally working simple SIP scenario). Is there any example of > such configuration? > 3) Let's imagine Proxy_2 in the above example, in the default config, it > looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" I > can specify the address of its upstream, Proxy_1, but how and where shall I > set the specifications of the downstream, Proxy_3? > I suspect it is somewhere in the Routing Logic block (line 449) but not > sure how it is exactly done. I see a "route(SIPOUT)" call for example but I > am not sure how and where the value of SIPOUT is modified. Is it a > representation of the downstream servers in the config file? > Thanks, > Alireza > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Hi, I have another update. The problem that I had is fixed now and it is fixed in an unexpected way! The fix seems to be in changing the SIPp default client scenario. In specific, once I change the UAC.xml in SIPp trunk to generate the "BYE" message's "To" field like this: To: sut <sip:[call_number]@[remote_ip]:[remote_port] instead of: To: sut <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] The problem is solved and Kamailio proxy forwards BYEs to the server. I am not sure why this is happening and if it more of a SIPp issue or a Kamailio issue. Neither I am sure of the meaning of the "[peer_tag_param]" in SIPp default client, but I thought it might be helpful to document the issue here in case that somebody else was searching the same issue. On Thu, Jun 19, 2014 at 8:42 PM, AliReza Khoshgoftar Monfared < khoshgoftar(a)gmail.com&gt; wrote: > Shedding some more light on the situation, here are some further facts I > could figure out: > 1) I did the same test without the proxy in between (UAC <--> UAS) purely > in SIPp, everything was fine, no dead call error > 2) With the Kamailio proxy in between, I can see that there is a problem > in call termination. The client is not receiving any "200 OK" for the > "BYE"s it is sending. > Is there anything wrong in my routing logic that is preventing these ACKs? > On Thu, Jun 19, 2014 at 4:09 PM, AliReza Khoshgoftar Monfared < > khoshgoftar(a)gmail.com&gt; wrote: >> Thanks very much Daniel, >>> This is solving a part of my issue for now. >> Here is the routing section of my configuration file (it may be good to >> share it once complete as a minimally working proxy): >>>> route{ >>> if (!mf_process_maxfwd_header("10")) { >>> sl_reply("483","Too Many Hops"); >>> break; >>> } >>> if (msg:len >= 4096 ) { >>> sl_reply("513", "Message too big"); >>>> } >>>> if (!method=="REGISTER") record_route(); >>>> if (!loose_route()) { >>> $du = "sip:10.236.214.86:5060"; >>> #setdsturi("sip:10.236.214.86:5060"); #second way to do >>> it >>> if (!t_relay()) { >>> sl_reply_error(); >>> } >>>> } >>> } >>>>>>>> I am trying to set up a kamailio proxy on an EC2 VM using this >> configurations and have been running a UAC and a UAS using SIPp on two >> other machines (UAC <--> Kamailio Proxy <--> UAS): >>> UAC >>>> sipp -sn uac -nr -r 1 -rp 1000 -d 0 -l 1 -p 5060 -trace_msg -i >>> SELF_IP(UAC) -rsa PROXY_IP:5060 UAS_IP:5060 >>>>> UAS: >>>> sipp -sn uas -d 0 -p 5060 -i SELF_IP(UAS) -rsa PRXY_IP:5060 -trace_msg >>>> It looks that messages go through, and are received by the server, but >> what I get back at the UAC (client) is a "dead call" error: >>> Last Error: Dead call 1-1734(a)10.140.34.188 (aborted at index 8), >>> receive... >>>>> Is there a specific meaning to this "dead call" error? Is there anything >> that my proxy is missing in its routing or does it have to do with the >> UAC/UAS configs? >>>> Thanks >> Alireza >>>> On Tue, Jun 17, 2014 at 5:11 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com&gt; wrote: >>>> Hello, >>>>> listen is to specify local ip address or network interface on which >>> kamailio should listen for sip traffic. >>>>> To send to an ip address there are couple of variants, in config file: >>>>> $ru = "sip:" + $rU + "@__NEXT_PROXY_IP__"; >>> t_relay(); >>> exit; >>>>> Or, if you don't want to change the r-uri, then use: >>>>> $du = "sip:__NEXT_PROXY_IP__"; >>> t_relay(); >>> exit; >>>>> Of course, you have to replace the __NEXT_PROXY_IP__ with the >>> appropriate value. >>>>> More dynamic option would be using dispatcher module. >>>>> Cheers, >>> Daniel >>>>>>> On 15/06/14 15:28, AliReza Khoshgoftar Monfared wrote: >>>>> Dear Kamailio users, >>>>> I am trying to set up a simple scenario as follows: >>>>> UAC --> Proxy_1 --> Proxy_2 --> Proxy_3 --> UAC >>>>> I am new to Kamailio and had the following basic questions that came >>> to my mind after reading the documentation an the default kamilio.cfg >>> config script: >>>>> 1) Is there any good example of the above scenario in which the UAC >>> and UAS (caller and callee) are modeled using SIPp and proxies are simple >>> kamailio instances run on different machines that simply forward the SIP >>> packets? >>>>> 2) In specific, I want the proxies to simply forward the messages to >>> their downstream (I do not care about registration or other operations and >>> look for a minimally working simple SIP scenario). Is there any example of >>> such configuration? >>>>> 3) Let's imagine Proxy_2 in the above example, in the default config, >>> it looks like that with modifying line 164, "listen=udp:10.0.0.10:5060" >>> I can specify the address of its upstream, Proxy_1, but how and where shall >>> I set the specifications of the downstream, Proxy_3? >>>>> I suspect it is somewhere in the Routing Logic block (line 449) but >>> not sure how it is exactly done. I see a "route(SIPOUT)" call for example >>> but I am not sure how and where the value of SIPOUT is modified. Is it a >>> representation of the downstream servers in the config file? >>>>>>> Thanks, >>> Alireza >>>>>>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>>>>>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>>>