October 2003 - sr-users

Re: [Serusers] rewrite & ACK forwarding problem
by lgfausak＠august.net 30 Apr '04

30 Apr '04

I have also had problems with getting the ACK back. I don't completely understand your configuration, you must allow for packets going both directions, right? Here is my config : route { # check to see if the message has been around too long # probably means that it is looping # if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; # # make sure the length of the message isn't too long! # if (len_gt( max_len )) { sl_send_reply("513", "Wow -- Message too large"); break; }; # # do the loose-routing thing, this is important! # if(loose_route()) { log(1,"doing top loose route"); t_relay(); break; }; # this is where I was dropping the ACKS. # I was simply dropping these, but they must be relayed # because they can be ACKs if(!(uri==myself)) { if(!t_relay()) { sl_reply_error(); break; }; break; }; This gets the ACKs through for me. By the way, I have this configured with Cisco ATAs, version 2.16. ---greg > >I have the same problem and posed it to the group yesterday ([Serusers] >Ignored 200 OK message.) So far the only workaround that I have found is to >use the rules in my gateway to rewrite the dialed digits before sending them >to the PSTN PRI, thus leaving the origianl URI intact for SIP >communications. > >One person told me that this is a bug in the Cisco ATA, but it happens on my >IPDialog phones also. It seems to me that the INVITE is being processed by >the SER dial rules and is rewritten, but the ACK is not. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/ > >----- Original Message ----- >From: "Alexander Mayrhofer" <axelm(a)nic.at> >To: <serusers(a)lists.iptel.org> >Sent: Friday, June 27, 2003 12:15 PM >Subject: [Serusers] rewrite & ACK forwarding problem > > >> >> Hi, >> >> we're running SER together with a PSTN Gateway. Before a call get's >> forwarded to the gateway, we are rewriting the request URI to make >> rewriting on the GW as simple as possible: >> >> route { >> ... >> strip(3); # +43xxx -> xxx >> prefix("0"); # xxx -> 0xxx >> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway >> route(1); >> break; >> ... >> >> SIP call flow looks like (record route enabled): >> >> (1) phone -> SER >> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0 >> >> (2) SER -> phone >> SIP/2.0 100 trying -- your call is important to us >> >> (3) SER -> GW >> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> (4) GW -> SER >> SIP/2.0 100 Trying >> >> (5) GW -> SER >> SIP/2.0 183 Session Progress >> >> (6) SER -> phone >> SIP/2.0 183 Session Progress >> >> (7) GW -> SER >> SIP/2.0 180 Ringing >> >> (8) SER -> phone >> SIP/2.0 180 Ringing >> >> (9) GW -> SER >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060> >> >> (10) SER -> phone >> SIP/2.0 200 OK >> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060> >> >> [ call established, we can talk, but ... ] >> >> (11) phone -> SER >> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0 >> >> --> Here starts the problem. That ACK (11) never gets forwarded to the >> Gateway, so after a few seconds, the GW starts over at (9). Those three >> packets (9-11) repeat a few times until GW runs into a timeout and drops >> the call. >> >> I have the impression that SER can't match the packet to the previous >> requests because of the rewritten URI. Is that correct? >> >> The only output at debug level 3 is: >> >> Warning: sl_send_reply: I won't send a reply for ACK!! >> >> Is that a routing goof somewhere in our scripts or is that a more >> generic problem? Is the problem that the warning indicates somehow >> related to the fact that the ACK is not being forwarded? >> >> Help appreciated. >> >> cheers >> >> axelm >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] If anyone can comment of asterisk problem
by John Foster 30 Jan '04

30 Jan '04

Hi List, i tried on asterisk list as well, but any of you can give any idea abt SIP authenticatin problem with Asterisk and SIP UA (i tried SIPPS and X-Lite) I could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321] type=friend username=12321 host=dynamic secret=ccarta context=default mailbox=1234,2345 ; Mailbox for message waiting indicator [77777] type=friend username=77777 host=dynamic secret=atracc context=default mailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 1 REGISTER Content-Length: 0 User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d" Content-Length: 0 to 192.168.100.66:5062 Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 2 REGISTER Content-Length: 0 Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="00000001",response="b8b1d7fc53eff354dfc31dfa3f800749" User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: <sip:12321@192.168.100.71>;expires=600 Date: Tue, 14 Oct 2003 13:46:14 GMT Content-Length: 0 to 192.168.100.66:5062 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:12321@192.168.100.66:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062> Contact: <sip:asterisk@192.168.100.71> Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.100.66:5062 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062>;tag=3b302259 Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 Content-Length: 0 8 headers, 0 lines --------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search

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RE: [Serusers] Latest STABLE build without using CVS
by Jiri Kuthan 20 Jan '04

20 Jan '04

At 07:39 PM 8/14/2003, Chad Brown wrote: >Perfect, > >Let me ask 2 quick follow-on questions... > >1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz >to get the latest patched and STABLE builds? That's the latest, most stable source, you need to compile it myself. There will be a new complete distribution by end of this month -- we are now waiting to run SER through the upcoming SIPIT to release it. >2. What location / version of the modules should I use when running >builds for this location? (Serweb, mysql, jabber, etc) All SER modules are included there. uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too. -jiri

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[Serusers] Radius accounting - radius_log_flag?
by Steven R. Bunin 16 Jan '04

16 Jan '04

Hi, I have radius authentication working thanks to Jan's help but I am still not receiving any accounting messages to my radius server. I found a mention of setting radius_log_flag but ser 0.9.11 tells me it isn't found in the "acc" module. How can I set invite and goodbye messages (or any others for that matter) to send information to my radius server for accounting purposes. I have read at leat 90% of the old messages but have yet to find an answer. Thanks in advance, Steve

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[Serusers] Xten and SER problem
by Jesus Rodriguez 17 Nov '03

17 Nov '03

Hello, I'm trying to register an Xten (1016 build) to SER and i always get a "401" response. Registering an ATA186 to the same SER with the same user and pass works with no problem (logs below Xten). We have tried SIPPS and registers ok too. Xten send the REGISTER, SER answers with a WWW-Authenticate challenge but Xten doesn't send anything, just sends a new REGISTER after few seconds: (c)2003 Xten Networks Inc. All rights reserved. Private build: 1016 SIP: 192.168.1.203:5060 RTP: 192.168.1.203:8000 NAT: 192.168.1.203 PXY#0: 192.168.1.2:5060 SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31483 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562000000000dded3e97ed7f89d3722ff12d3704a884",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" SEND >> 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2> Contact: "JesusR" <sip:1000@192.168.1.203:5060> Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER Expires: 500 User-Agent: X-Lite build 1016 Content-Length: 0 RECEIVE << 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.203:5060 From: <sip:1000@192.168.1.2> To: <sip:1000@192.168.1.2>;tag=d907c037823644515dfe0ede38ca9976.d844 Call-ID: 854E48C84C104A5DB92CE4A65ABC2406(a)192.168.1.2 CSeq: 31485 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c562200000000e06a238bd1858769f2e46b11d8d837f1",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13588 req_src_ip=192.168.1.203 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" The same action works ok with ATA186: [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 1 REGISTER WWW-Authenticate: Digest algorithm=MD5,nonce="3f1c579a00000000aa6fd2432e429f54e3 7e75981218834b",realm="voztelecom.net" Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13590 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" [0]Reg Resp 401; Unauthorized [0]REGISTER Retry 0 [0:0] Tx Msg to 192.168.1.2:5060 REGISTER sip:192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone> Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;expires=60 User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Authorization: Digest username="4070",realm="voztelecom.net",nonce="3f1c579a0000 0000aa6fd2432e429f54e37e75981218834b",uri="sip:192.168.1.2",response="fe07b8be83 0305cca7a28bf35c112270" Content-Length: 0 [0]Rx Msg from 192.168.1.2:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.202:5060 From: <sip:4070@192.168.1.2;user=phone>;tag=403901681 To: <sip:4070@192.168.1.2;user=phone>;tag=d907c037823644515dfe0ede38ca9976.9cf4 Call-ID: 2622180140(a)192.168.1.202 CSeq: 2 REGISTER Contact: <sip:4070@192.168.1.202:5060;user=phone;transport=udp>;q=0.00;expires=6 0 Server: Sip EXpress router (0.8.10 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.2:5060 "Noisy feedback tells: pid=13589 req_src_ip=192.16 8.1.202 in_uri=sip:192.168.1.2 out_uri=sip:192.168.1.2 via_cnt==1" Thanks in advance. Saludos JesusR. ------------------------------- Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------

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[Serusers] how to install ser and configure it with voicemail
by kelvin 06 Nov '03

06 Nov '03

I have install ser-0.8.11-0.i386.rpm, mysql and serweb. My sip phone are all window messenger. When someone contact a user who is not on-line, you get this message:- phone(a)192.168.1.3 has declined your request to have a voice conversation. When I login 'phone' to serwweb as a subscriber, under missed calls is always empty & message store too. How to make it ring first and after a period go to voicemail if phone(a)192.168.1.3 is not online? I went through the document but got confuse. Do you have a example ser.cfg script that works. 2. Through the web documents there are 2 vm softwares sems-0.1.0-0.i386.rpm & ortp-0.6.2.tar.gz. which is the right one? 3. if vm is required must you reinstall ser? If yes, what to change in the makefile? Help! Thanks!

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[Serusers] RADIUS authentication: draft-sterman-aaa-sip
by Jiri Kuthan 03 Nov '03

03 Nov '03

Unfortunately, there is now no standard for use of RADIUS along with SIP. SER users leveraging the combination of these two technologies are left with implementation of expired internet drafts. There are some chances that the IETF community revitalizes the document. Thus, I would appreciate hearing if any of the active RADIUS/SIP/SER users have had any issues with the RADIUS authentication in SER, which is based on draft-sterman-aaa-sip. Thanks, -jiri -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] New release of seradmin v .03e
by Xten India 02 Nov '03

02 Nov '03

Hai, We Xten released a newer version (seradmin v .03e) of SERAdmin. Xten is building this for the ser community and we would very much like your feedback so that we can build you a better product The Source code is available at http://developer.berlios.de/projects/seradmin SERAdmin is a GUI interface between SIP Express Router (SER) and a SER administrator.SERAdmin has an intuitive look and feel. SERAdmin provides control over many SER tasks such as: Start, Stop, Pause, Re-start, Monitor SER, Add User, Change Password and EmailId Delete User, Add Alias , Edit Alias etc. With this we can also use FIFO commands,Access controls,User Location. The objective of the SERAdminv03e is to support the ser-0.8.11 version also. This latest SER Administration Application handles both ser-0.8.10 and ser-0.8.11. What is new?. The following commands available in this release ------------------------------------------------ Ping(uri) - Pinging a URI Cisco_restart(uri) - Restart a Cisco Phone arg - Arguments of SER pwd - Present Working Directory t_uac_dlg - Initiate a Transaction you can download the components from http://developer.berlios.de/projects/seradmin For any feedback & help pls. contact xten_india(a)yahoo.com Regards, Team, Xten_india --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software

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[Serusers] Outbound proxy
by Peter Hudec 31 Oct '03

31 Oct '03

hello, this is maybe a bit silly question, but. Outbound proxy is needed, when UA is behind FW. What is the function of this proxy (how can this help the UA to make calls) and what kind of software can do this. Can SER be also outbound proxy, or I have to use something else ? hudecof -- mail: [phudec(a)postel.sk] www: [http://www.postel.sk] cellular: [+421 02 50203166] icq: [99518783] gpg: [http://hudecof.net/data/hudecof.gpg]

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[Serusers] Gentoo
by danny adams 31 Oct '03

31 Oct '03

Is this latest version of Ser already intergrated on release of latest version......in Gentoo. Danny --------------------------------- Post your free ad now! Yahoo! Canada Personals

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[Serusers] SEMS
by Alessio Focardi 31 Oct '03

31 Oct '03

Hi, I need some documentation for SEMS, havent found none by now. I have it running and working but I would like to explore more possibilities, for instance: Message playback via SIP client or/and Storing the voice message in a folder then sending an email with a http link to it instead of using attach method of/and Compressing the message using MP3/Ogg and so on ... Also: can SEMS be used to provide voice mail messages insted of result codes for failed calls ? Something like "the user is busy", "no such user". I tested addaline.com service, that runs on SER, and they do this. Any help will be appreciated! TNX -- Best regards, Alessio mailto:alessiof@interconnessioni.it

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[Serusers] Re: One Way Audio call
by Raymond May 31 Oct '03

31 Oct '03

Hi, I have just installed SER on my Linux Debian server. It seems to work just great for users on the Internet calling each other ( 2 way Audio is successful ). However if someone calls me ( I am behind my SER server ) they can hear me but I can't hear them ( 1 way Audio ONLY ). Based on all the docs I've read it seems to have something to do with NAT and RTP ports. My Phone connected to LAN --- SER Server connected to my LAN on one interface and to DSL Provider on the other ---- DSL Service Provider. I am also doing NAT/PAT on SER server. Below is my Topology and ser.cfg. Appreciate any feedback I get. My Phone --- SER --- Internet --- Users that register with my SER server. SER.CFG File # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic 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"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

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RE: [Serusers] Outbound proxy
by Klaus Darilion 31 Oct '03

31 Oct '03

Hi Peter! Not every SIP proxy can deal with clients behind a firewall/NAT box. Therefore it is useful to us an outbound proxy which is known to be capable of NATed clients. This proxy will send keep alive messages to your client to keep the pinhole in your firewall open and rewrites the IP addresses in the SIP messages of the client behind the FW (or NAT box). yes. ser can be an outbound proxy. every proxy can be a outbound proxy. An outbound proxy is not a special proxy type. Outbound proxy just means that the SIP UA client always sends the SIP messages to this proxy instead of contacting the other participant directly. regards, klaus > -----Original Message----- > From: Peter Hudec [mailto:phudec@postel.sk] > Sent: Saturday, November 01, 2003 12:21 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Outbound proxy > > > hello, > > this is maybe a bit silly question, but. > > Outbound proxy is needed, when UA is behind FW. > What is the function of this proxy (how can this help the UA to make > calls) and what kind of software can do this. > > Can SER be also outbound proxy, or I have to use something else ? > > hudecof > > -- > mail: [phudec(a)postel.sk] www: [http://www.postel.sk] > cellular: [+421 02 50203166] icq: [99518783] > gpg: [http://hudecof.net/data/hudecof.gpg] > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] Fwd: SIP Tutorial at IEEE CCNC
by Jiri Kuthan 31 Oct '03

31 Oct '03

FYI -- there will be a SIP tutorial in the upcoming CCNC conference, hold by Charles Baker. -jiri >I will be presenting a half-day tutorial on SIP at the IEEE Consumer >Communications and Networking Conference in Las Vegas on Thursday, January >8. This is a new conference which is positioned to overlap the Consumer >Electronics Show. See < www.CCNC2004.org > [...] > Dr. H. Charles Baker, P.E., > Professional Web Site: http://engr.smu.edu/~hbaker

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[Serusers] t_uac_dlg() bug ?
by Jim Burwell 31 Oct '03

31 Oct '03

Hi, I was trying to get click-to-dial working from SERweb and it was failing for unclear reasons. I then experimented with the "ctd.sh" script, which does the same thing as the SERweb PHP code. I found that it would fail during the 'dummy invite' stage, returning an odd error if I used a URI which used a SIP domain/realm, but would work fine if I used the IP address of the SIP proxy as the domain. i.e.: ./ctd.sh sip:joebob@mydomain.com sip:billybob@mydomain.com <- fails ./ctd.sh sip:joebob@1.2.3.4 sip:billybob@mydomain.com <- succeeds The reply from SER would be "500 fifo_uac: no mem for hf block" . The log file would have these errors: ERROR: mk_proxy: could not resolve hostname: "mydomain.com" ERROR: uri2proxy: bad host name in URI <sip:joebob@mydomain.com> ERROR: uri2sock: Can't create a dst proxy ERROR: get_hf_block: send_sock failed ERROR: fifo_uac: no mem for hf block So it appears that SER didn't understand that the URI's domain was itself, even though there's a host alias line in the ser.cfg file listing "mydomain.com" as an alias for this host. It's as if it thinks it needed to relay the transaction to another SIP router, and look up an IP to do so. If "mydomain.com" were placed in /etc/hosts as an alias for the system SER was running on, or if a DNS SRV entry for the SIP domain were created pointing to the SER system, voila, the "sip:user@domain" form would work just fine. Is this a bug ? I think SER isn't checking SER host aliases against the SIP domain if the SIP request comes in via the fifo t_uac_dlg() command. I'm using the SER release 0.8.11-r1, aka the "stable CVS" release. TIA, Jim -- +---------------------------------------------------------------------------+ | Jim Burwell - Sr. Systems/Network/Security Engineer, JSBC | +---------------------------------------------------------------------------+ | "I never let my schooling get in the way of my education." - Mark Twain | | "UNIX was never designed to keep people from doing stupid things, because | | that policy would also keep them from doing clever things." - Doug Gwyn | | "Cool is only three letters away from Fool" - Mike Muir, Suicyco | | "..Government in its best state is but a necessary evil; in its worst | | state an intolerable one.." - Thomas Paine, "Common Sense" (1776) | +---------------------------------------------------------------------------+ | Email: jimb(a)jsbc.cc ICQ UIN: 1695089 | +---------------------------------------------------------------------------+ | Reply problems ? Turn off the "sign" function in email prog. Blame MS. | +---------------------------------------------------------------------------+

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[Serusers] SerWeb
by jason 31 Oct '03

31 Oct '03

I have Ser installed and running on debian, I have mysql installed with the ser module. I can see the table the module created. I am getting this error message and not sure what it is, When trying the link..127.0.0.1/html/admin/index.php *Warning*: Unable to include '/prolog.html' - request execution failed in */var/www/html/page.php* on line *46* iptel.org User Management *Warning*: Unable to include '/separator.html' - request execution failed in */var/www/html/page.php* on line *48* *Warning*: Unable to include '/epilog.html' - request execution failed in */var/www/html/page.php* on line *82 Any ideas? *

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[Serusers] Warning - Noisy Feedback when trying to register from a messenger to SER
by Madhuri garigipati 31 Oct '03

31 Oct '03

Hello All, I am trying to register a user from a Messenger Application. I am able to register that user successfully and I am recieving a SIP/2.0 200 OK message. And I am able to see the registered user using the serctl utility. But I am getting the following warning message: Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1" Can some one explain the reason for this warning and how to remove it? Find the complete send and recieve message trace below: ---- Sending ------- REGISTER sip:rajesh@12.213.181.254 SIP/2.0 Via: SIP/2.0/UDP 67.161.15.238 From: madhu <sip:madhu@67.161.15.238> To: rajesh <sip:rajesh@12.213.181.254> Call-ID: 522193516(a)67.161.15.238 Expires:10000 m: madhu <sip:madhu@67.161.15.238> CSeq: 1 REGISTER ---- Receiving ----- SIP/2.0 200 OK Via: SIP/2.0/UDP 67.161.15.238 From: madhu <sip:madhu@67.161.15.238> To: rajesh <sip:rajesh@12.213.181.254>;tag=b27e1a1d33761e85846fc98f5f3a7e58.d4f5 Call-ID: 522193516(a)67.161.15.238 CSeq: 1 REGISTER Contact: <sip:madhu@67.161.15.238>;q=0.00;expires=10000 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1" Thanks, Madhuri. --------------------------------- Do you Yahoo!? Exclusive Video Premiere - Britney Spears

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Re[3]: [Serusers] Ser and Voicemail
by Alessio Focardi 31 Oct '03

31 Oct '03

Hello Jiri, Wow ... voicemail is working now: TNX ! I can ear "something" with my Xlite client but voice is pretty distorted, I suppose that I have to convert the default announcement to another sample rate/frequency/codec. Unfortunately nothing is recorded, here is a trascript of voicemail session, any help will be appreciated ... 200 OK 58383:45571714 00000BD46BBF2446 Contact: <sip:300001@interconnessioni.it:5090> Content-Type: application/sdp . v=0 o=username 0 0 IN IP4 127.0.0.1 s=session c=IN IP4 127.0.0.1 t=0 0 m=audio 30410 RTP/AVP 3 a=rtpmap:3 gsm/8000 . > (3163) DEBUG: openReplyFifo (AmRequest.cpp:292): openReplyFifo start (3163) DEBUG: openReplyFifo (AmRequest.cpp:313): creating reply fifo filehandle (3163) DEBUG: openReplyFifo (AmRequest.cpp:334): opening reply fifo (3163) DEBUG: write_to_fifo (AmRequest.cpp:635): Write to fifo: completed (3163) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded (3163) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> (3163) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> (3163) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> (3163) DEBUG: run (AmSession.cpp:115): Sending Rtp data to 212.141.166.135/8000 (3163) DEBUG: onSessionStart (AnswerMachine.cpp:65): starting sending announce...(/sems/audio/default_en.wav) (3163) DEBUG: trying to read WAV file (3163) DEBUG: tag = <RIFF> (3163) DEBUG: file size = <129710> (3163) DEBUG: tag = <WAVE> (3163) DEBUG: tag = <fmt > (3163) DEBUG: chunk_size = <18> (3163) DEBUG: fmt = <01> (3163) DEBUG: channels = <1> (3163) DEBUG: rate = <8000> (3163) DEBUG: bits/sample = <16> (3163) DEBUG: tag = <fact> (3163) DEBUG: chunk size = <4> (3163) DEBUG: tag = <data> (3163) DEBUG: chunk size = <129660> (3163) DEBUG: play (AmRtpStream.cpp:244): AmRtpStream::play: using payload 3 (3163) ERROR: pcm16_2_gsm: number of blocks should be integral (block size = 320) (3163) DEBUG: onSessionStart (AnswerMachine.cpp:67): ... announce reaches its end. (3163) DEBUG: onSessionStart (AnswerMachine.cpp:73): starting recording message...(/tmp/4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135.wav) (3163) DEBUG: record (AmRtpStream.cpp:267): AmRtpStream::record: using payload 3 ... (3163) DEBUG: onSessionStart (AnswerMachine.cpp:76): ... finished recording message (3163) DEBUG: onSessionStart (AnswerMachine.cpp:82): recorded data size: 0 (3163) DEBUG: sadSession (AmSession.cpp:274): searching for session to destroy (hash=4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135569566049,sess_key=00000BD46BBF2446) (3163) DEBUG: destroySession (AmSession.cpp:313): sess_it == sessions.end() (3028) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded (3028) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> (3028) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> (3028) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> (3030) DEBUG: run (AmSession.cpp:213): Session cleaner starting its work (3030) DEBUG: run (AmSession.cpp:227): session 80e6740 has been destroyed' (3030) DEBUG: run (AmSession.cpp:244): Session cleaner finished -- Best regards, Alessio mailto:alessiof@interconnessioni.it

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[Serusers] Pdt module troubles with accounting
by Todd Wright 30 Oct '03

30 Oct '03

Hi, Im currently trying to set SER 0.8.11 up and am having a problem with the pdt module which translates addresses based on a numeric prefix. When the accounting module tries to write its record to the database (mysql), the INSERT query is truncated just after the address that has been translated by the pdt module. This causes mysql to fail to insert the record and give an error about query syntax. Other accounting records for untranslated addresses are inserted properly with no error. Is this a known issue? I suspect the translated address is being null terminated by the pdt module but Im not enough of a code hacker to tell for sure. -- _--_|\ --------- Todd Wright -- wylie(a)geekasylum.org -------- / \ ICQ: 9589981 YIM: mvs38j \_.--._* <--- http://www.geekasylum.org/~wylie/ v Mobile: +61-403-796-001 Ph: +61-2-9699-1746 ----------------------------------------------------------------

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[Serusers] Question about an adaptation with SER
by Ivan Dario Salazar 30 Oct '03

30 Oct '03

Hi... Thanks a lot for your help I have a question... I have tens of telephony lines in a university that all they work with PSTN... so i wish i could do something with SER to manage the phone calls that they do, and this way save money...is it possible, what i need(Hardware, Software). Thankyou so much.. IVAN DARIO _________________________________________________________ http://www.latinmail.com. Gratuito, latino y en espa�ol.

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Re: [Serusers] RTP Proxy and SER
by Stephen Miles 30 Oct '03

30 Oct '03

THanks for that Andrei, all sorted. One last thing, getting a call from the PSTN to go to a soft phone registered with ser. I have tried a few route clauses like if method==invite lookup location etc but it doesn't seem to be working. Is there a simple thing I am missing here? Thanks Stephen Andrei Pelinescu-Onciul wrote: > On Oct 29, 2003 at 17:55, Stephen Miles <smiles(a)callplus.co.nz> wrote: > >>Hi Andrei, >> >>I have grabbed the latest cvs version and have compiled it and seems to >>be running ok. I now get an error 5(3681) ERROR: extract_mediaip: no >>`c=' in SDP when I try to place a call from a soft phone on my laptop >>through the ser proxy running on the proxy box. Any ideas? > > > > The 'c=' in SDP specifies the IP address you use for receiving media. > It shouldn't be missing (see RFC 2327 "SDP: Session Description Protocol"). > > >>Also is there a way to get ser to forward/relay registers to another server? >>We have a SIP server (broadsoft) setup and I need all the register and >>call info, and calls to be routed from ser to the broadsoft server. > > > Yes, you can do this stateless or statefull. > > stateless: > forward("server", port); > > statefull: > make sure the tm module is loaded: > > t_relay_to_udp("server", port); > >>Is it possible to set ser up as just a proxy where it takes the client >>request and then proxies it to our other sip server? > > > Yes, see above and set ser+nathelper box as outbound proxy for you ua. > > You should also record route everything, to make sure all the > subsequent requests in the dialogs pass through your proxy (if you are > using it for nat traversal you want this). > For a record routing example see the default ser.cfg (on cvs sources in > etc/ser.cfg) and copy the loose_route & record_route stuff. > > > More info about configuring ser: http://www.iptel.org/ser/admin.html > (a little bit outdated) > > > Andrei ----------------------------------------------------------------------------------------------- This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments.

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Re: [Serusers] SIP Instant Messaging UA
by Jiri Kuthan 30 Oct '03

30 Oct '03

4.6 and 4.7 are mostly ok, but 5 breaks the specification quite violently. kphone seems quite ok today for linux. I'm not aware of a solid IM SIP UA for Windows other than elderly Messenger versions. There are some conversations about building a reasonable client, but they are still in very early stage. -jiri At 11:44 AM 10/28/2003, dhiraj.2.bhuyan(a)bt.com wrote: >Hello List, > >Can anyone recommend some SIP Instant Messaging UA (linux and windows) that follows the specification word by word? >I tried MSN Messenger 4.6 and 5 - both appears to have some problems (version 5 tries to send a MESSAGE with the INVITE method and gets back a BYE in response!!!!!!!!!!!!!). > >Thanks in advance, > >Dhiraj Bhuyan >Security Research Engineer >BT Exact > >Tel: +44 1473 643932 >Mob: +44 7962 012145 >Email: dhiraj.2.bhuyan(a)bt.com > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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RE: [Serusers] SER does not start with MySQL
by Jiri Kuthan 30 Oct '03

30 Oct '03

At 07:49 PM 10/30/2003, Vo, Tony wrote: >Workaround is change the DIR=... lines in serctl to path to ser. >>>Tony>> I've tried this and no luck..... well, then try something else. perhaps watching the serctl execution could help (replace leading line #!/bin/sh with #!/bin/sh -x) -jiri

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RE: [Serusers] SER does not start with MySQL
by Vo, Tony 30 Oct '03

30 Oct '03

Hi Jiri, Please read my comments to your questions below..... -----Original Message----- From: Jiri Kuthan [mailto:jiri@iptel.org] Sent: Wednesday, October 29, 2003 7:03 PM To: Vo, Tony; serusers(a)lists.iptel.org Subject: Re: [Serusers] SER does not start with MySQL I suppose ser would not start without mysql either. The reason is presumably serctl's way to find SER executable. It tries to use the same directory for SER as where serctl lives, certainly a problem with compiled versions and possibly a problem in some distributions. Where is your 'ser' and 'serctl' installed? >>Tony>> /usr/sbin Which distribution are you using? >>Tony>> I installed version 8.11 for RedHat 8.0 (ser-0.8.11-0.i386.rpm). SER works fine without loading mysql module. Problem only occurs when trying to load the mysql module. Workaround is change the DIR=... lines in serctl to path to ser. >>Tony>> I've tried this and no luck..... I introduced (hopefuly) more helpful error output to serctl. Thanks for reporting, -jiri At 03:20 AM 10/30/2003, Vo, Tony wrote: >content-class: urn:content-classes:message >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C39E8C.7267B729" > >Hi, >SER will not start if I am to configure SER to run using MySQL module. I followed the instructions in the HowTo guides. I currently have MySQL version 4.0.16 installed and is running. The mysql.so module is in the correct directory as specified in the config file. When I run the 'serctl start' script, I would get only this response; > >[root@localhost modules]# serctl start > >Starting SER : cat: /var/run/ser.pid: No such file or directory >started pid() > >Here's an excerpt from my config file (ser.cfg). Thanks for your advice(s); > ># ># ># $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ ># ># simple quick-start config script ># > ># ----------- global configuration parameters ------------------------ > >#debug=3 # debug level (cmd line: -dddddddddd) >#fork=yes >#log_stderror=no # (cmd line: -E) > >/* Uncomment these lines to enter debugging mode >debug=7 >fork=no >log_stderror=yes >*/ > >check_via=no # (cmd. line: -v) >dns=no # (cmd. line: -r) >rev_dns=no # (cmd. line: -R) >port=5060 >#children=4 >fifo="/tmp/ser_fifo" > ># ------------------ module loading ---------------------------------- > ># Uncomment this if you want to use SQL database >loadmodule "/usr/lib/ser/modules/mysql.so" > >loadmodule "/usr/lib/ser/modules/sl.so" >loadmodule "/usr/lib/ser/modules/tm.so" >loadmodule "/usr/lib/ser/modules/rr.so" >loadmodule "/usr/lib/ser/modules/maxfwd.so" >loadmodule "/usr/lib/ser/modules/usrloc.so" >loadmodule "/usr/lib/ser/modules/registrar.so" > ># Uncomment this if you want digest authentication ># mysql.so must be loaded ! >loadmodule "/usr/lib/ser/modules/auth.so" >loadmodule "/usr/lib/ser/modules/auth_db.so" > ># ----------------- setting module-specific parameters --------------- > ># -- usrloc params -- > >#modparam("usrloc", "db_mode", 0) > ># Uncomment this if you want to use SQL database ># for persistent storage and comment the previous line >modparam("usrloc", "db_mode", 2) >modparam("usrloc", "db_url", "sql://ser:spirent@localhost/ser") > ># -- auth params -- ># Uncomment if you are using auth module ># >modparam("auth_db", "calculate_ha1", 1) ># ># If you set "calculate_ha1" parameter to yes (which true in this config), ># uncomment also the following parameter) ># >modparam("auth_db", "password_column", "password") > >#database location for auth module >modparam("auth_db", "db_url", "sql://serro:spirent@localhost/ser") > ># -- rr params -- ># add value to ;lr param to make some broken UAs happy ># modparam("rr", "enable_full_lr", 1) > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re[3]: [Serusers] Ser and Voicemail
by Jiri Kuthan 30 Oct '03

30 Oct '03

please keep the mailing list cc-ed -- there are buddies who do understand SEMS unlike myself. -jiri At 05:09 PM 10/30/2003, Alessio Focardi wrote: >Hello Jiri, > >Wow ... voicemail is working now: TNX ! > >I can ear "something" with my Xlite client but voice is pretty >distorted, I suppose that I have to convert the default announcement >to another sample rate/frequency/codec. > >Unfortunately nothing is recorded, here is a trascript of voicemail session, >any help will be appreciated ... > >200 >OK >58383:45571714 >00000BD46BBF2446 >Contact: <sip:300001@interconnessioni.it:5090> >Content-Type: application/sdp >. >v=0 >o=username 0 0 IN IP4 127.0.0.1 >s=session >c=IN IP4 127.0.0.1 >t=0 0 >m=audio 30410 RTP/AVP 3 >a=rtpmap:3 gsm/8000 >. > >> >(3163) DEBUG: openReplyFifo (AmRequest.cpp:292): openReplyFifo start >(3163) DEBUG: openReplyFifo (AmRequest.cpp:313): creating reply fifo filehandle >(3163) DEBUG: openReplyFifo (AmRequest.cpp:334): opening reply fifo >(3163) DEBUG: write_to_fifo (AmRequest.cpp:635): Write to fifo: completed >(3163) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded >(3163) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> >(3163) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> >(3163) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> >(3163) DEBUG: run (AmSession.cpp:115): Sending Rtp data to 212.141.166.135/8000 >(3163) DEBUG: onSessionStart (AnswerMachine.cpp:65): starting sending announce...(/sems/audio/default_en.wav) >(3163) DEBUG: trying to read WAV file >(3163) DEBUG: tag = <RIFF> >(3163) DEBUG: file size = <129710> >(3163) DEBUG: tag = <WAVE> >(3163) DEBUG: tag = <fmt > >(3163) DEBUG: chunk_size = <18> >(3163) DEBUG: fmt = <01> >(3163) DEBUG: channels = <1> >(3163) DEBUG: rate = <8000> >(3163) DEBUG: bits/sample = <16> >(3163) DEBUG: tag = <fact> >(3163) DEBUG: chunk size = <4> >(3163) DEBUG: tag = <data> >(3163) DEBUG: chunk size = <129660> >(3163) DEBUG: play (AmRtpStream.cpp:244): AmRtpStream::play: using payload 3 >(3163) ERROR: pcm16_2_gsm: number of blocks should be integral (block size = 320) >(3163) DEBUG: onSessionStart (AnswerMachine.cpp:67): ... announce reaches its end. >(3163) DEBUG: onSessionStart (AnswerMachine.cpp:73): starting recording message...(/tmp/4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135.wav) >(3163) DEBUG: record (AmRtpStream.cpp:267): AmRtpStream::record: using payload 3 >... > >(3163) DEBUG: onSessionStart (AnswerMachine.cpp:76): ... finished recording message >(3163) DEBUG: onSessionStart (AnswerMachine.cpp:82): recorded data size: 0 >(3163) DEBUG: sadSession (AmSession.cpp:274): searching for session to destroy (hash=4FC9AB89-F383-412E-930B-E1694B82AAFB(a)212.141.166.135569566049,sess_key=00000BD46BBF2446) >(3163) DEBUG: destroySession (AmSession.cpp:313): sess_it == sessions.end() >(3028) DEBUG: getReturnCode (AmRequest.cpp:380): response from Ser: 200 fifo_t_reply succeeded >(3028) DEBUG: getReturnCode (AmRequest.cpp:401): r_uri= <> >(3028) DEBUG: getReturnCode (AmRequest.cpp:404): cmd.next_hop= <> >(3028) DEBUG: getReturnCode (AmRequest.cpp:405): cmd.route= <> >(3030) DEBUG: run (AmSession.cpp:213): Session cleaner starting its work >(3030) DEBUG: run (AmSession.cpp:227): session 80e6740 has been destroyed' >(3030) DEBUG: run (AmSession.cpp:244): Session cleaner finished > > > > > >-- >Best regards, > Alessio mailto:alessiof@interconnessioni.it -- Jiri Kuthan http://iptel.org/~jiri/

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sr-users October 2003