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37455 discussions

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

4 6

[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

2 2

[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

4 8

[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

2 1

[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

3 2

[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

1 0

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