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Re: [Users] openser and rtpproxy on multihomed host
by Alexander 'Leo' Bergolth 05 Jun '07

05 Jun '07

On 06/05/2007 05:17 PM, raviprakash sunkara wrote: > Hi Alexander > Can use fix_nated_sdp("flag"); and also force_rtp_proxy("flags"); I'm already using force_rtp_proxy() (without arguments) and it already rewrites the "c=" address in the sdp information but it uses the wrong interface address. (The openser-server/rtpproxy has two IP adresses in the same subnet.) Since I know the address which is used by the clients or contacting the server, I could hardcode the address as the second parameter to force_rtp_proxy to force the use of this address but I was looking for a cleaner solution. I'd like to use the address where the SIP request came in for both RTP streams. Maybe I can use the pseudo-variable $Ri (Received IP address) as the second parameter to force_rtp_proxy? This should force the use of the correct IP address in the "c=" field and will hopefully persuade the client to use this IP address as the destination address for its RTP data. But how do I tell rtpproxy to send its RTP data from this address? Thanks, --leo > On 6/5/07, Alexander Bergolth <leo(a)strike.wu-wien.ac.at> wrote: >> >> Hi! >> >> I'm experiencing problems using openser and rtpproxy on a multihomed box >> (IP addresses are 137.208.89.45 and 137.208.89.65). >> >> The client (firewalled) contacts the server via the first IP address >> (137.208.89.45) but rtpproxy sometimes uses it's second IP >> (137.208.89.65) in the "c=" SDP-information. (See below.) >> However it _sends_ RTP packets from the _first_ IP address. Those >> packets will never pass through the firewall, of course, since the >> clients outgoing RTP traffic is sent to another IP address than the >> incoming traffic is hitting the firewall from. >> >> Is there a way to tell rtpproxy to use the same IP-address both for the >> c= SDP header and as source address? >> >> -------------------- 8< -------------------- >> ## >> T 2007/06/05 15:57:36.041537 137.208.89.45:5060 -> >> 137.208.90.164:3099[AP] >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/TCP >> 137.208.90.164:56136 >> ;branch=z9hG4bK-d87543-a107a77a6311422d-1--d87543-;rport=3099. >> From: "Alexander Bergolth"<sip:30001@wu-wien.ac.at>;tag=5b6d7b31. >> To: "996050"<sip:996050@wu-wien.ac.at>;tag=as2e40807f. >> Call-ID: MTVjYzlhMDdjMjAwODExZDEyNzk5M2IwN2UxNjhlYTc.. >> CSeq: 2 INVITE. >> User-Agent: Asterisk PBX. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Supported: replaces. >> Contact: <sip:996050@137.208.89.135>. >> Content-Type: application/sdp. >> Content-Length: 283. >> . >> v=0. >> o=root 6978 6978 IN IP4 137.208.89.135. >> s=session. >> c=IN IP4 137.208.89.65. >> t=0 0. >> m=audio 35064 RTP/AVP 0 8 101. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=silenceSupp:off - - - -. >> a=ptime:20. >> a=sendrecv. >> a=nortpproxy:yes. >> -------------------- 8< -------------------- >> >> -------------------- 8< -------------------- >> 15:57:35.814136 IP 137.208.90.164.3099 > 137.208.89.45.5060: P >> 4236850032:4236851080(1048) ack 224540451 win 64256 >> 15:57:35.814147 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 1048 >> win 2003 >> 15:57:35.814351 IP 137.208.89.45.5060 > 137.208.90.164.3099: P >> 1:550(549) ack 1048 win 2003 >> 15:57:35.814430 IP 137.208.89.45.5060 > 137.208.90.164.3099: P >> 550:1264(714) ack 1048 win 2003 >> 15:57:35.814850 IP 137.208.90.164.3099 > 137.208.89.45.5060: . ack 1264 >> win 64177 >> 15:57:35.915829 IP 137.208.90.164.3099 > 137.208.89.45.5060: P >> 1048:1418(370) ack 1264 win 64177 >> 15:57:35.955948 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 1418 >> win 2003 >> 15:57:36.017309 IP 137.208.90.164.3099 > 137.208.89.45.5060: P >> 1418:2672(1254) ack 1264 win 64177 >> 15:57:36.017320 IP 137.208.89.45.5060 > 137.208.90.164.3099: . ack 2672 >> win 2003 >> 15:57:36.017397 IP 137.208.89.45.5060 > 137.208.90.164.3099: P >> 1264:1813(549) ack 2672 win 2003 >> 15:57:36.018604 IP 137.208.89.45.5060 > 137.208.90.164.3099: P >> 1813:2388(575) ack 2672 win 2003 >> 15:57:36.019170 IP 137.208.90.164.3099 > 137.208.89.45.5060: . ack 2388 >> win 64256 >> 15:57:36.041454 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP, >> length 172 >> 15:57:36.041537 IP 137.208.89.45.5060 > 137.208.90.164.3099: P >> 2388:3200(812) ack 2672 win 2003 >> 15:57:36.046899 IP 137.208.90.164.10677 > 137.208.89.65.35065: UDP, >> length 132 >> 15:57:36.061155 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP, >> length 172 >> 15:57:36.080128 IP 137.208.90.164.10676 > 137.208.89.65.35064: UDP, >> length 172 >> 15:57:36.081157 IP 137.208.89.45.35064 > 137.208.90.164.10676: UDP, >> length 172 >> [...] >> -------------------- 8< -------------------- >> >> Thanks in advance, >> --leo >> >> P.S.: I'm using openser-1.2 and rtpproxy-cvs20070222. >> >> -- >> e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at >> fax ::: +43-1-31336-906050 >> location ::: Computer Center | Vienna University of Economics | Austria >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > > > -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

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[Serusers] SER Error
by Rjey Nomer 05 Jun '07

05 Jun '07

Hi, Can anyone help me with my problem. I configure SER and Asterisk on different machine. Objective are calling SER Client ---> Asterisk Client and Asterisk Client ---> SER client. Problem is when I call SER client using my Asterisk client it can successfully ring the ser client but when I pickup the phone, it doesn't have any voice. Error occured on my messages log. ======================== Jun 5 23:47:14 ser ser[7263]: Warning: sl_send_reply: I won't send a reply for ACK!! Jun 5 23:47:14 ser ser[7263]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Jun 5 23:47:14 ser ser[7266]: ERROR: mk_proxy: could not resolve hostname: "@192.168.1.41" Jun 5 23:47:14 ser ser[7266]: ERROR: uri2proxy: bad host name in URI <sip:@192.168.1.41;ftag=as3add8ffb;lr=on> Jun 5 23:47:14 ser ser[7266]: ERROR: t_forward_nonack: failure to add branches ======================== Thanks in advanced! Regards, Rjey ____________________________________________________________________________________ Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php

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[Users] New sip:wizard features
by Andreas Granig 05 Jun '07

05 Jun '07

All, I want to announce a bunch of new features for sip:wizard, the OpenSER configuration generator. Beside some improvements, the focus for this round was on conditional call forwarding. For a detailed description, see: http://www.sipwise.com/news To try them out, visit: http://www.sipwise.com/wizard Best regards, Andreas

1 0

[Users] avp_check in OpenSER 1.1
by Carsten Bock 05 Jun '07

05 Jun '07

Hi Everybody, I have a question regarding avp_check() and reverse-DNS-Lookups: I wanted to do the following in the routing logic (simplified): loose_route(); if (avp_check("$dd","re/192\.168.123.(1|2)/g")) { The idea was to allow flexible re-routing to another proxy in case of loose-routing. One proxy would be the alias of the other, and so far, everything was fine. What happened now: We had a customer, who's IP could not be "reverse-lookuped" properly, the DNS-Server ran into an timeout. After we removed the code-snipplet from the routing, everything was fine (again); except this functionality drawback. I believe, in OpenSER 1.2 i could replace it with the following, to avoid the reverse-lookups: if (($dd == 192.168.123.1) || ($dd == 192.168.123.2)) { Am i right? Would this work? Can i somehow avoid the reverse-lookup in OpenSER 1.1? Thank you everybody, Carsten

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[Users] 200-OK and sendto() problem
by Andreas Granig 05 Jun '07

05 Jun '07

Hi, I'm going crazy with this problem, so maybe anyone here can see any error or can give a hint how to debug this. I've two clients behind (the same) NAT. INVITE goes thru, so does 100 and 180. But the 200 isn't relayed because of the following syslog error: ./sbin/openser[23933]: ERROR: udp_send: sendto(sock,0x81816a8,814,0,0xb5b7f168,16): Operation not permitted(1) ./sbin/openser[23933]: msg_send: ERROR: udp_send failed I can't find any error (the two Via headers are exactly the same), so here is the 180 which is relayed without problems, and below the 200, which causes the error: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.19.176.11;branch=z9hG4bK6351.1b30ad43.0 Via: SIP/2.0/UDP 192.168.123.100:2083;received=83.65.24.161;branch=z9hG4bK-kq6qyyeowiei;rport=12499 From: "Foo Bar" <sip:foo1000@linguin.org>;tag=xuu6a2tp6b To: <sip:2000@linguin.org;user=phone>;tag=2091138726 Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE CSeq: 2 INVITE Server: Cisco ATA 186 v3.1.1 atasip (040629A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 84.19.176.11;branch=z9hG4bK6351.1b30ad43.0 Via: SIP/2.0/UDP 192.168.123.100:2083;received=83.65.24.161;branch=z9hG4bK-kq6qyyeowiei;rport=12499 Record-Route: <sip:84.19.176.11;lr=on;ftag=xuu6a2tp6b;nat=yes> From: "Foo Bar" <sip:foo1000@linguin.org>;tag=xuu6a2tp6b To: <sip:2000@linguin.org;user=phone>;tag=2091138726 Call-ID: 3c27e5729c40-csdi3p954zyn@snom360-0004132306CE CSeq: 2 INVITE Contact: <sip:foo2000@192.168.123.103:5060;transport=udp> Server: Cisco ATA 186 v3.1.1 atasip (040629A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Supported: replaces Content-Length: 204 Content-Type: application/sdp v=0 o=foo2000 11828 11828 IN IP4 192.168.123.103 s=ATA186 Call c=IN IP4 192.168.123.103 t=0 0 m=audio 16384 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Any ideas? Regards, Andreas PS: this happens with both OpenSER 1.1 and 1.2

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[Users] OpenSER Not able to send Notify
by kiran.bkumar＠wipro.com 05 Jun '07

05 Jun '07

Hi All, Finally I was able to get the presence server up and running with the openims and also able to do some basic tests with xlite. But the problem I am seeing is when I change the status of a user (for ex from available to busy), it sends a publish event to openser and now I expect the openser to trigger a Notify to the contacts. But this does not seem to be happening and I see some errors as below, 1(5293) SIP Request: 1(5293) method: <PUBLISH> 1(5293) uri: <sip:kiran@mkd149633.wipro.com> 1(5293) version: <SIP/2.0> 1(5293) parse_headers: flags=2 1(5293) Found param type 232, <branch> = <z9hG4bK4ac2.743db705.0>; state=16 1(5293) end of header reached, state=5 1(5293) parse_headers: Via found, flags=2 1(5293) parse_headers: this is the first via 1(5293) After parse_msg... 1(5293) preparing to run routing scripts... 1(5293) parse_headers: flags=100 1(5293) Found param type 232, <branch> = <z9hG4bK4ac2.bd17e9a7.0>; state=16 1(5293) end of header reached, state=5 1(5293) parse_headers: Via found, flags=100 1(5293) parse_headers: this is the second via 1(5293) Found param type 234, <received> = <10.117.2.44>; state=6 1(5293) Found param type 232, <branch> = <z9hG4bK-d87543-52058d7c8c660725-1--d87543->; state=6 1(5293) Found param type 235, <rport> = <5186>; state=16 1(5293) end of header reached, state=5 1(5293) parse_headers: Via found, flags=100 1(5293) DEBUG:maxfwd:is_maxfwd_present: value = 15 1(5293) DEBUG: t_newtran: T on entrance=0xffffffff 1(5293) parse_headers: flags=ffffffffffffffff 1(5293) DEBUG:parse_to:end of header reached, state=10 1(5293) DBUG:parse_to: display={"Kiran"}, ruri={sip:kiran@mkd149633.wipro.com} 1(5293) DEBUG: get_hdr_field: <To> [40]; uri=[sip:kiran@mkd149633.wipro.com] 1(5293) DEBUG: to body ["Kiran"<sip:kiran@mkd149633.wipro.com> ] 1(5293) get_hdr_field: cseq <CSeq>: <95> <PUBLISH> 1(5293) DEBUG: get_hdr_body : content_length=449 1(5293) found end of header 1(5293) parse_headers: flags=78 1(5293) t_lookup_request: start searching: hash=11428, isACK=0 1(5293) DEBUG: RFC3261 transaction matching failed 1(5293) DEBUG: t_lookup_request: no transaction found 1(5293) parse_headers: flags=ffffffffffffffff 1(5293) PRESENCE:handle_publish: SIP-If-Match found 1(5293) PRESENCE:handle_publish: existing etag = a.1181035469.5298.17 1(5293) PRESENCE: handle_publish: 'expires' found 1(5293) PRESENCE: handle_publish: lexpire= 60 1(5293) PRESENCE: handle_publish: 'To' header ALREADY PARSED: <sip:kiran@mkd149633.wipro.com> 1(5293) parse_headers: flags=ffffffffffffffff 1(5293) PRESENCE:publ_send200ok: send 200OK reply 1(5293) PRESENCE:publ_send200ok: etag= a.1181035469.5298.17 - len= 20 1(5293) parse_headers: flags=ffffffffffffffff 1(5293) check_via_address(10.154.20.45, 10.154.20.45, 0) 1(5293) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= kiran,[domain]= mkd149633.wipro.com for [event]= presence returned no result 1(5293) PRESENCE:query_db_notify: Could not get subs_dialog from database 1(5293) PRESENCE:update_presentity: Could not send Notify 1(5293) DEBUG: cleanup_uac_timers: RETR/FR timers reset 1(5293) DEBUG: add_to_tail_of_timer[2]: 0xb609b268 (6756) 1(5293) DEBUG:tm:UNREF_UNSAFE: after is 0 1(5293) DEBUG:destroy_avp_list: destroying list (nil) 1(5293) receive_msg: cleaning up 5(5299) DEBUG: timer routine:2,tl=0xb609b268 next=(nil), timeout=6756 5(5299) DEBUG: wait_handler : removing 0xb609b220 from table 5(5299) DEBUG: delete transaction 0xb609b220 5(5299) DEBUG: wait_handler : done Can you please help me on this. Let me know if you need some log files/traces. Do I require to do some configuration changes or any additions to the database for this. Thanks, Kiran. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Users] pb with install the Openser and Module SQL
by li wang 05 Jun '07

05 Jun '07

Dear all, I got an error while using openser to set up the presence server in my Debian post: I have installed openser_1.2.0-0_i386.deb, but i don't find mysql.so in /usr/lib/openser/modules. So i go install the openser-mysql-module_1.2.0-0_i386.deb. It is installed but with a problem of dependance : openser-mysql-module depends libmysqlclient14 while libmysqlclient15 installed in my post. however, i can run the script openser_mysql to set up the data base. But when i restart OpenSER after edit the /etc/openser.cfg to support mysql, i got error like this: pc7:/usr/lib/openser/modules# /etc/init.d/openser restart Not starting openser: invalid configuration file! 0(15900) ERROR: load_module: could not open module </usr/lib/openser/modules/mysql.so>: libmysqlclient.so.14: cannot open shared object file: No such file or directory 0(15900) parse error (44,13-14): failed to load module ERROR: bad config file (1 errors) 0(15900) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy But libmysqlclient14 is obsolete in the Debian repositories, there is only libmysqlclient15. Don't you have encounter the similar problem? Please help me. Thanks! My linux distribution is Linux 2.6.18-4-686. li _________________________________________________________________ Gagnez des pc Windows Vista avec Live.com http://www.image-addict.fr/

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Re: [Serusers] SER and Asterisk
by Rjey Nomer 05 Jun '07

05 Jun '07

I already define the hostname of both machines and I created also an internal DNS wherein I define the following: ================================== # dig -t SRV _sip._udp.rjey.ph ; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.rjey.ph ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 13180 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 5 ;; QUESTION SECTION: ;_sip._udp.rjey.ph. IN SRV ;; ANSWER SECTION: _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 asterisk.rjey.ph. _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 ser.rjey.ph. ;; AUTHORITY SECTION: rjey.ph. 10800 IN NS ns1.rjey.ph. rjey.ph. 10800 IN NS ns2.rjey.ph. ;; ADDITIONAL SECTION: ser.rjey.ph. 10800 IN A 192.168.1.41 asterisk.rjey.ph. 10800 IN A 192.168.1.247 ns1.rjey.ph. 10800 IN A 192.168.1.7 ;; Query time: 46 msec ;; SERVER: 192.168.1.7#53(192.168.1.7) ;; WHEN: Tue Jun 5 18:32:58 2007 ;; MSG SIZE rcvd: 292 ============================ What else should I do???? Thanks, Rjey Date: Tue, 05 Jun 2007 07:24:27 -0400 From: "SIP" <sip(a)arcdiv.com> Add to Address Book Add Mobile Alert To: "Rjey Nomer" <rjeynomer(a)yahoo.com> CC: serusers(a)iptel.org Subject: Re: [Serusers] SER and Asterisk This means you can't resolve 192.168.1.247 from your SER server. Try adding an entry for it into /etc/hosts to see if you can bypass DNS. N. Rjey Nomer wrote: > Hi all, > > Hope someone can help me with this. > > The main objective are to make Asterisk and SER > communicate with each other. Call SER--> Asterisk and > Asterisk--> SER. > > I used the example configuration for pstn as base on > all the docs, it is how we can make ser and asterisk > work. As instructed on the docs, we just need to add > the IP address of the PSTN gateway on the trusted > database of our SER, in this case, asterisk is our > PSTN gateway. > > =================== > mysql> insert into trusted values > ("192.168.1.247","any","^sip:.*$"); > =================== > > On the asterisk (trixbox) server, below are the > configuration I defined: > > Outgoing Setting: > Trunk Name: serout > Peer Details: > ==================== > allow=all > dtmfmode=rfc2833 > host=192.168.1.41 > insecure=no > type=peer > ==================== > > I can ring any number on my SER Server using any > number on my Asterisk, problem is when I pick-up the > phone I cannot hear voice and according to the logs as > listed below, it is hanging-up and something like > "Unresolvable destination". > > > SER NGREP RESULT > #ngrep -n 5060 -d eth0 3242194ngrep -n 5060 -d eth0 > 3242194 > ================================= > U 192.168.1.41:5060 -> 192.168.1.247:5060 > SIP/2.0 478 Unresolvable destination (478/TM)..Via: > SIP/2.0/UDP > 192.168.1.247:5060;branch=z9hG4bK580c7aec;rport=5060..F > rom: "3000" > <sip:3000@192.168.1.247>;tag=as7693144e..To: > <sip:3242194@192.168.1.41>;tag=419e8..Call-ID: > 5784a2681edd513 > c59c77e20512b499d@192.168.1.247..CSeq: 103 > INVITE..Server: Sip EXpress router (0.9.3 > (i386/linux))..Content-Length: 0.. > Warning: 392 192.168.1.41:5060 "Noisy feedback > tells: pid=9413 req_src_ip=192.168.1.247 > req_src_port=5060 in_uri=sip:3 > 242194@192.168.1.6:52961;user=phone > out_uri=sip:3242194@192.168.1.6:52961;user=phone > via_cnt==1".... > ================================= > > Asterisk Logs > ================================= > -- Executing NoOp("SIP/3000-08fd48b8", "CallerID set > to "3000" <3000>") in new stack > -- Executing Set("SIP/3000-08fd48b8", > "GROUP()=OUT_2") in new stack > -- Executing GotoIf("SIP/3000-08fd48b8", "0?108") > in new stack > -- Executing Set("SIP/3000-08fd48b8", > "DIAL_NUMBER=3242194") in new stack > -- Executing Set("SIP/3000-08fd48b8", > "DIAL_TRUNK=2") in new stack > -- Executing AGI("SIP/3000-08fd48b8", > "fixlocalprefix") in new stack > -- Launched AGI Script > /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Could not parse > /etc/asterisk/localprefixes.conf > -- AGI Script fixlocalprefix completed, returning > 0 > -- Executing Set("SIP/3000-08fd48b8", > "OUTNUM=3242194") in new stack > -- Executing Set("SIP/3000-08fd48b8", > "custom=SIP/Serout") in new stack > -- Executing GotoIf("SIP/3000-08fd48b8", "0?16") > in new stack > -- Executing Dial("SIP/3000-08fd48b8", > "SIP/Serout/3242194|120|r") in new stack > -- Called Serout/3242194 > -- SIP/Serout-08fda208 is ringing > -- SIP/Serout-08fda208 answered SIP/3000-08fd48b8 > -- Attempting native bridge of SIP/3000-08fd48b8 > and SIP/Serout-08fda208 > -- Got SIP response 478 "Unresolvable destination > (478/TM)" back from 192.168.1.41 > == Spawn extension (macro-dialout-trunk, s, 14) > exited non-zero on 'SIP/3000-08fd48b8' in macro > 'dialout-trunk' > == Spawn extension (macro-dialout-trunk, s, 14) > exited non-zero on 'SIP/3000-08fd48b8' > > ================================= > > Can anyone help me resolve this problem. > > Thanks in advanced. > > Rjey > > > > ____________________________________________________________________________________ > Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. > http://farechase.yahoo.com/ > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > ____________________________________________________________________________________ Don't get soaked. Take a quick peak at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather

2 2

[Users] error 407
by amit 05 Jun '07

05 Jun '07

Hi All, when i send a reinvite request proxy give a error 407 (Proxy Authentication Required). why it is .......... Thanks amit

1 0

[Serusers] SER and Asterisk
by Rjey Nomer 05 Jun '07

05 Jun '07

Hi all, Hope someone can help me with this. The main objective are to make Asterisk and SER communicate with each other. Call SER--> Asterisk and Asterisk--> SER. I used the example configuration for pstn as base on all the docs, it is how we can make ser and asterisk work. As instructed on the docs, we just need to add the IP address of the PSTN gateway on the trusted database of our SER, in this case, asterisk is our PSTN gateway. =================== mysql> insert into trusted values ("192.168.1.247","any","^sip:.*$"); =================== On the asterisk (trixbox) server, below are the configuration I defined: Outgoing Setting: Trunk Name: serout Peer Details: ==================== allow=all dtmfmode=rfc2833 host=192.168.1.41 insecure=no type=peer ==================== I can ring any number on my SER Server using any number on my Asterisk, problem is when I pick-up the phone I cannot hear voice and according to the logs as listed below, it is hanging-up and something like "Unresolvable destination". SER NGREP RESULT #ngrep -n 5060 -d eth0 3242194ngrep -n 5060 -d eth0 3242194 ================================= U 192.168.1.41:5060 -> 192.168.1.247:5060 SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP 192.168.1.247:5060;branch=z9hG4bK580c7aec;rport=5060..F rom: "3000" <sip:3000@192.168.1.247>;tag=as7693144e..To: <sip:3242194@192.168.1.41>;tag=419e8..Call-ID: 5784a2681edd513 c59c77e20512b499d@192.168.1.247..CSeq: 103 INVITE..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0.. Warning: 392 192.168.1.41:5060 "Noisy feedback tells: pid=9413 req_src_ip=192.168.1.247 req_src_port=5060 in_uri=sip:3 242194@192.168.1.6:52961;user=phone out_uri=sip:3242194@192.168.1.6:52961;user=phone via_cnt==1".... ================================= Asterisk Logs ================================= -- Executing NoOp("SIP/3000-08fd48b8", "CallerID set to "3000" <3000>") in new stack -- Executing Set("SIP/3000-08fd48b8", "GROUP()=OUT_2") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?108") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_NUMBER=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_TRUNK=2") in new stack -- Executing AGI("SIP/3000-08fd48b8", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/3000-08fd48b8", "OUTNUM=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "custom=SIP/Serout") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?16") in new stack -- Executing Dial("SIP/3000-08fd48b8", "SIP/Serout/3242194|120|r") in new stack -- Called Serout/3242194 -- SIP/Serout-08fda208 is ringing -- SIP/Serout-08fda208 answered SIP/3000-08fd48b8 -- Attempting native bridge of SIP/3000-08fd48b8 and SIP/Serout-08fda208 -- Got SIP response 478 "Unresolvable destination (478/TM)" back from 192.168.1.41 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8' ================================= Can anyone help me resolve this problem. Thanks in advanced. Rjey ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/

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