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37472 discussions

[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

> > Hi all, > i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: > > A -------------------------- SER ---------------------------------- B > SIPp SIPp > > > i have a problem using SER : > I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. > > the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. > > i m putting below my config file ser.cfg > > i have also some questions: > > 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. > 2/ also i have noticed that when using the command : serctl moni , i got the following message: > root@amar-desktop:~# serctl moni > [: 1074: ==: unexpected operator > [: 1074: 0: unexpected operator > > despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. > > 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. > > i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. > > > Amar > > > > > _________________________________________________________________ > You live life beyond your PC. So now Windows goes beyond your PC. > http://clk.atdmt.com/MRT/go/115298556/direct/01/ _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

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[Serusers] problem with SER not proxying messages
by amar mahmoud 03 Nov '08

03 Nov '08

Hi all, i am new to use SER. i want to build a test bed using SER as proxy server and SIPp as user agents to make some measurements as follows: A -------------------------- SER ---------------------------------- B SIPp SIPp i have a problem using SER : I did registration for the user agents with the server and i can see them in location table in SER database, when A start a call to B the server respond to the INVITE with 100 trying then can't relay the message to B until A gets 408 request timeout , it seems that the server cannot retrieve the contact of B from the database. the Request URI i am using is sip:B@192.168.165.18 which is the IP of the server and i tried also using alias like ser.org so it became sip:B@ser.org but still have the same problem. i m putting below my config file ser.cfg i have also some questions: 1/ the domain field of location table in database is empty, so is this normal situation, actually i did set it manually to my domain. 2/ also i have noticed that when using the command : serctl moni , i got the following message: root@amar-desktop:~# serctl moni [: 1074: ==: unexpected operator [: 1074: 0: unexpected operator despite that ser is starting by using the command : serctl start, and i can see it is starting and even there is no error in /var/log/syslog. 3/ is there a way to view the sip messages traversing the server, i mean to monitor it., and is /var/log/syslog the only place to see the logs. i would be thankful if someone can help me with that cause i have task to do and i got stuck beside having very limited time. Amar _________________________________________________________________ You live life beyond your PC. So now Windows goes beyond your PC. http://clk.atdmt.com/MRT/go/115298556/direct/01/

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[Kamailio-Users] irc devel meeting
by Daniel-Constantin Mierla 03 Nov '08

03 Nov '08

Hello, being the time for a new IRC devel meeting, I propose Thursday, Nov 6, 2008, at 15:00UTC, on channel #kamailio, server irc.freenode.net The focus is to sync the developers related the work since the last major release, roadmap to next major release as well as minor releases scheduling. Please add to the agenda at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] re mote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

________________________________ Monday, November 3, 2008 3:28:30 AM arif.zaman Wrote: > Did you check the location table whether callee is registered before > forwarding any call? that is strange that I see in the table location it, registered : +----+----------+------------------------------------------+------------------------+------------------------------+ | id | username | contact | received | user_agent | +----+----------+------------------------------------------+------------------------+------------------------------+ | 1 | 122 | sip:122@192.168.0.60:5065 | sip:190.184.35.4:5065 | Linksys/SPA942-5.2.8 | | 2 | 201 | sip:201@192.168.11.30:5060;transport=udp | sip:192.168.11.30:5060 | Grandstream BT200 1.1.4.18 | | 4 | 202 | sip:202@192.168.10.30:5064;transport=udp | sip:190.212.5.155:5064 | Grandstream GXP2020 1.1.6.16 | +----+----------+------------------------------------------+------------------------+------------------------------+ regardsss rickygm ________________________________

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[Kamailio-Users] INVITE accounting problem
by Kaeman Chris 03 Nov '08

03 Nov '08

Hello, I have a problem in accounting INVITE. When the call is connected with duration the INVITE, ACK, BYE are accounted in db. But the problem is when the call is disconnected before it is actually attempting to connect. i am getting only CANCEL accounted whereas INVITE is not accounted for the cancelled call. I have the xlogs showing INVITE and CANCEL both for this call only no accounting is performed for INVITE. following is the config for INVITE & CANCEL; if (is_method("INVITE")) { setflag(1); setflag(2); # MISSED if (allow_trusted()) { rewritehostport("xx.xxx.xy.xyz:5060"); route(1); } else { sl_send_reply("401", "Unauthorized"); route(1); exit; } } # CANCEL processing if (is_method("CANCEL")) { setflag(1); setflag(3); if (t_check_trans()) t_relay(); exit; } t_check_trans(); Please let me know what should be changed so that for the Call that is cancelled before it actually connects can have INVITE also accounted along w/ CANCEL. Thanks, KChris. Connect with friends all over the world. Get Yahoo! India Messenger at http://in.messenger.yahoo.com/?wm=n/

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[Kamailio-Users] restrict calls from gateway to some extensions
by muhammad akl 03 Nov '08

03 Nov '08

Hey all , I want to make some ACL for the INVITES come from a PSTN-GW to certain users on my OpenSER server. for ex: I have the following extensions on my OpenSER server : 11001 11002 | | | | 11111 What I want to implement is something similar to this : Caller <-------> PSTN-GW <------->OpenSER <----> (complete this call to this extension ) 11001 Caller <-------> PSTN-GW <------->OpenSER <----> (reject this call for this user ) 11002 and so on for other extensions . any way to implement this feature in openser ?

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[Kamailio-Users] remote clients don't contact other remote clients
by Ricky Gutierrez 03 Nov '08

03 Nov '08

Hello list, I have another question and a very strange detail, I have a couple of connected remote clients to my kamailio 1.3.2 and mediaproxy 1.9.1, when the client 122 calls to the client 202 this very contacts or, if I call a minute I contact him very well, but if I call to the client 122 after some minutes I don't contact with that client and the jumps me to the voicemail . I see in the chart location the address ip of the UAC that it can be happening? regards rickygm 122 UAC <-> NAT <-> internet==== kamailio 1.3.2 / mediaproxy ==== internet <-> ADSL <-> NAT <-> UAC 202 public ip: x.x.x.x.x interface: any filter: (ip) and ( port 5060 ) # U +0.890415 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000286 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 INVITE Proxy-Authenticate: Digest realm="190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.197976 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK1eff324647d9fef9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=329cfeaa6ded039da25ff8cbb8668bd2.ea16 Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 390 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.073962 190.212.35.223:5064 -> 190.184.22.152:5060 INVITE sip:122@190.184.22.152 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: replaces, timer, path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 238 v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.10.30 t=0 0 m=audio 5004 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.003823 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK7288d140481a11e9;rport=5064;received=190.212.35.223 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 # U +0.000053 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.485936 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.999801 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +1.999612 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999219 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.999220 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +4.001208 190.184.22.152:5060 -> 190.184.35.4:5065 INVITE sip:122@192.168.0.60:5065 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.0 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +3.599535 190.184.22.152:5060 -> 192.168.11.1:5070 INVITE sip:u122@192.168.11.1:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: replaces, timer, path Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 240 P-hint: route(3)|setflag7,forcerport,fix_contact P-hint: inbound->inbound P-hint: Route[6]: mediaproxy v=0 o=202 8000 8001 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 # U +0.000362 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152> Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Length: 0 # U +0.000283 190.184.22.152:5070 -> 190.184.22.152:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.1;received=190.184.22.152 Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 15334 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/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 # U +0.002152 190.184.22.152:5060 -> 190.212.35.223:5064 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.30:5064;rport=5064;received=190.212.35.223;branch=z9hG4bK7288d140481a11e9 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:u122@190.184.22.152:5070> Content-Type: application/sdp Content-Length: 289 P-hint: onreply_route|usemediaproxy v=0 o=root 9069 9069 IN IP4 190.184.22.152 s=session c=IN IP4 190.184.22.152 t=0 0 m=audio 35020 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/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 # U +0.236892 190.212.35.223:5064 -> 190.184.22.152:5060 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bK0fd7eaebf7c6f6c6 Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@192.168.10.30:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 # U +0.000248 190.184.22.152:5060 -> 190.184.22.152:5070 ACK sip:u122@190.184.22.152:5070 SIP/2.0 Record-Route: <sip:190.184.22.152;lr=on;ftag=86ecf1029bb72cb7;nat=yes> Via: SIP/2.0/UDP 190.184.22.152;branch=z9hG4bK5397.fb4a49d3.3 Via: SIP/2.0/UDP 192.168.10.30:5064;received=190.212.35.223;branch=z9hG4bK0fd7eaebf7c6f6c6 From: <sip:202@190.184.22.152>;tag=86ecf1029bb72cb7 To: <sip:122@190.184.22.152>;tag=as6ee3962f Contact: <sip:202@190.212.35.223:5064;transport=udp> Supported: path Proxy-Authorization: Digest username="202", realm="190.184.22.152", algorithm=MD5, uri="sip:122@190.184.22.152", nonce="490eb5583ea2f71836195c92a7e7df100eec3075", response="aa82c7aaba810bcc0b46d7e7ea442603" Call-ID: 8976ac66491970eb(a)192.168.10.30 CSeq: 391 ACK User-Agent: Grandstream GXP2020 1.1.6.16 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 P-hint: LR|fixcontact,setflag6 exit 40 received, 0 dropped

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 02 Nov '08

02 Nov '08

Hi list, always looking for as solving my audio problem with mediaproxy asterisk and openser, there will be some form of telling to the openser that when he comes from the from sip:asterisk@192.168.10.1:5070 that doesn't use the mediaproxy or the onreply_route[1] , SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK36b7f619;rport=5070 Record-Route: <sip:192.168.10.1;lr=on;ftag=as42edbc9b;nat=yes> From: "asterisk" <sip:asterisk@192.168.10.1:5070>;tag=as42edbc9b To: <sip:113@192.168.10.1>;tag=6d45d2188218c8ef Call-ID: 5a1c60382f18bd832fa3bdc54dc6ab13(a)192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: <sip:113@192.168.10.30:5062;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 212 P-hint: Onreply-route - fixcontact P-hint: onreply_route|usemediaproxy v=0 o=113 8000 8000 IN IP4 192.168.10.30 s=SIP Call c=IN IP4 192.168.1.64 t=0 0 m=audio 35064 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 making other tests if I change this form the onreply_route[1], I have audio in the openser extension, but the one that this behind the pstn doesn't have audio or he doesn't listen to me onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } my best regardss rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

________________________________ Friday, October 31, 2008 2:59:58 AMDaniel-Constantin Mierla wrote: > do you have to bridge the RTP between the two network interfaces? >I do not know mediaproxy, but rtpproxy (nathelper module) has bridge mode >that can be used in such scenarios. Hi Daniel, I don't have a bridge with the two net cards, I had read some rtpproxy that can be used in bridge mode, but I didn't find alone examples some comments, but he gave me an error and I could not make it work this problem has me sad... and I am not seeing an exit.. regards .. rickygm

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Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy
by Ricky Gutierrez 31 Oct '08

31 Oct '08

Thank you Luciano for your answer but doesn't it still work me, do the calls enter me of the pstn and don't I listen anything... where I can look for? , there will be some of form so that the IP of where the calls come from asterisk does not use mediaproxy, somebody has an example. any help is appreciated.. thank you rickygm ________________________________ From: luzango mfupe <luzango.mfupe(a)gmail.com> Sent: Friday, October 31, 2008 5:19:25 AM Subject: Re: [Kamailio-Users] I don't have asterisk audio to openser - mediaproxy Hi Ricky, I think you should increase the mediaproxy natping interval to 30 and then add use_media_proxy() line after you setflag(6). check below:--- ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); USE MEDIA PROXY HERE---> use_media_proxy(); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } On Fri, Oct 31, 2008 at 5:21 AM, Ricky Gutierrez <xserverlinux(a)yahoo.com> wrote: I show you here my openser.cfg my best regards rickygm loadmodule "nathelper.so" loadmodule "mediaproxy.so" modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("nathelper","rtpproxy_sock", "") modparam("nathelper","natping_interval", 0) modparam("nathelper","received_avp", "$avp(i:42)") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!is_method("REGISTER")) { if(nat_uac_test("19")){ record_route(";nat=yes"); } else { record_route(); }; }; if (has_totag()) { # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; if (method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 if(nat_uac_test("19") || search("^Route:.*;nat=yes")){ append_hf("P-hint: LR|fixcontact,setflag6\r\n"); fix_contact(); setbflag(6); }; route(1); } else { sl_send_reply("404", "Not here"); } exit; } ##Loose_route packets if (loose_route()) { if(!has_totag()){ sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; }; route(1); }; #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); if (method=="REGISTER"){ route(2); } else { route(3); }; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { ; if (method=="INVITE" || method=="ACK") { ; end_media_session(); ; }; # sl_reply_error(); }; exit; } route[2] { # # -- Register request handler -- # if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); fix_nated_register(); force_rport(); }; save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { # # -- INVITE request handler -- # if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if (client_nat_test("3")) { append_hf("P-hint: route(3)|setflag7,forcerport,fix_contact\r\n"); setbflag(7); force_rport(); fix_contact(); }; if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(4); route(1); exit; } consume_credentials(); # Verify aliases lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound route(10); } else { # -- Inbound to outbound route(11); }; } else { # From an external domain -> do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { #-- Outbound to inbound route(12); } else { # -- Outbound to outbound route(13); }; }; } route[6] { # # -- NAT handling -- # if (isbflagset(6) || isbflagset(7)) { append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); }; } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table append_hf("P-hint: inbound->inbound \r\n"); if (uri=~"^sip:[2346578][0-9]{6}@.*") { if (is_user_in("credentials", "local")){ # rewritehostport("192.168.10.1:5070"); route(6); route(4); exit; } else { sl_send_reply("403", "No tienes permiso para llamadas locales"); exit; }; }; if (!lookup("location")) { if (does_uri_exist()) { revert_uri(); prefix("u"); rewritehostport("192.168.10.1:5070"); route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; }; sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } route[4] { rewritehostport("192.168.10.1:5070"); route(1); } ### aqui se detecta cunado una llamada es no disponible , ocupada o falla failure_route[1] { if (t_was_cancelled()) { exit; }; if (t_check_status("486")) { revert_uri(); prefix("b"); xlog("L_ERR","Stepped into the 486 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); xlog("L_ERR","Stepped into the 480 ruri=<$ru>"); rewritehostport("192.168.10.1:5070"); append_branch(); route(1); exit; }; } onreply_route[1] { # #-- On-replay block routing -- # if (client_nat_test("1")) { append_hf("P-hint: Onreply-route - fixcontact \r\n"); fix_nated_contact(); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (search("^Content-Type:[ ]*application/sdp")) { append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); }; }; exit; } -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

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