sr-users September 2003

sr-users@lists.kamailio.org

86 participants
209 discussions

[Serusers] Implementing Voicemail
by Steve Dolloff 09 Sep '03

09 Sep '03

I was looking at an old thread regarding this. http://lists.iptel.org/pipermail/serusers/2003-July/002003.html 1) How do I know when to direct it to this portion of the code? After x rings? 2) Am I mistaken in thinking that there is a voicemail system built in? Do I need to install a separate voicemail system? If not, how do I configure the voice mail system? If so, can anyone recommend one? Thanks, Stephen

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[Serusers] bearer capability not implemented
by Steve Dolloff 08 Sep '03

08 Sep '03

I am using a Cisco ATA 186 and a Cisco AS5350 in conjunction with a SER SIP server. When I try to place calls from the ATA to the PSTN, I am getting the following error. Is there a command that I can issue on the Gateway to provide more information for debug? Thanks, Stephen Total call-legs: 2 1284 : 46480131hs.94 +-1 +0 pid:1 Originate 13125551212 dur 00:00:00 tx:0/0 rx:0/0 41 (bearer capability not implemented) Telephony 3/0:D (180): tx:0/0/0ms None noise:0dBm acom:0dBm 1284 : 46480129hs.95 +-1 +3 pid:700 Answer 256 dur 00:00:00 tx:0/0 rx:0/0 41 (bearer capability not implemented) IP 209.242.10.153:16384 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw U 66.155.138.5:5060 -> 209.242.10.153:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 209.242.10.153;branch=z9hG4bK7b35.fb17 329.0,SIP/2.0/UDP 216.145.234.84:5060..From: <sip:256@209.242.10.153;user=p hone>;tag=741484739..To: <sip:18475551212@209.242.10.153;user=phone>;tag=1B C0FF10-3C3..Date: Thu, 06 Jan 2000 09:20:30 GMT..Call-ID: 1667923238(a)216.14 5.234.84..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1 INVITE..Content-Length : 0.... # U 66.155.138.5:5060 -> 209.242.10.153:5060 SIP/2.0 501 Not Implemented..Via: SIP/2.0/UDP 209.242.10.153;branch=z9hG4bK 7b35.fb17329.0,SIP/2.0/UDP 216.145.234.84:5060..From: <sip:256@209.242.10.1 53;user=phone>;tag=741484739..To: <sip:18475551212@209.242.10.153;user=phon e>;tag=1BC0FF10-3C3..Date: Thu, 06 Jan 2000 09:20:30 GMT..Call-ID: 16679232 38@216.145.234.84..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 1 INVITE..Conte nt-Length: 0....

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[Serusers] mysql table create logon error
by t d 08 Sep '03

08 Sep '03

while attempting logon at http://mydomain.com/admin/index,php with username and password , i get an "error in SQL query, line: 24" following the HowTo guide, i ran /usr/sbin/ser_mysql.sh create (version 1.38.2.6) when i query "select * from user;" i got error "no database selected" but when i connect ser; show tables; i can see all the table fields. on /var/www/html/config.php, i have configured the database connection user to "root" password "mysqlpassword" DB name "ser". any insights greatly appreciated. tim __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com

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[Serusers] route one call to two UA's?
by Jev 08 Sep '03

08 Sep '03

Hi all, Im replacing an old analog PBX with cisco ATA's, ser and a vega 50 BRI. Currently, on the old system, when some one rings the PSTN number, it goes through and rings two different phones (one upstairs, one downstairs), and the first person to answer the phone gets the call. Now, I need to replicate the behavior with ser, and the ata 186's. Has any one attempted this type of behavior? Is it within the capability of sip? Thanks, -Jev

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[Serusers] serweb/user_interface not returning answer
by Philippe Petit 08 Sep '03

08 Sep '03

Hi everybody, I'm having troubles with the Serweb 0.8.11 /user_interface/login. I have no problem with the standard admin login (after some corrections...) but urls in the user_interface folder don't return any result. I just got endless timer pointer. Any help? thank you Philippe

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Re: [Serusers] dropping call after ~15 sec "I won't send a reply for ACK!!"
by Jev 08 Sep '03

08 Sep '03

Hi, Adding record_route(); to my routing block solved this problem, calls work great now! thank you for the pointer :) -Jev Greg Fausak wrote: > Howdy, > > I had this problem as well. I believe it was due to the fact > that I didn't to record routing, and the ACK didn't get back. > My calls lasted exactly 9 seconds. > > Anyway, it is really handy to do call tracing to debug > these sorts of things. If you have a unix machine, tcpdump -w, > or a windows machine can use ethereal. You save the sniffer > session and then process it with 'sipscenario'. It makes a very > easy to look at call flow, which will probably highlight the > problem. > > Of course to do sniffing you have to have a machine that sees all > of the packets. > > ---greg > Greg Fausak > Addaline.com, Inc. > > > >>Hi, >> >>I'm setting up a vega 50 BRI with ser, and I can make in/out bound >>calls, quality is great! >> >>I have a problem, when I dial from a sip phone to a land line, my call >>gets set up, and works great, but after approx 15 seconds it drops the >>call. Below is what I get in the logs from ser. >> >> >> 0(29026) DEBUG: is_maxfwd_present: max_forwards header not found! >> 0(29026) DEBUG: add_param: tag=4063815580 >> 0(29026) end of header reached, state=29 >> 0(29026) parse_headers: flags=256 >> 0(29026) find_first_route(): No Route headers found >> 0(29026) loose_route(): There is no Route HF >> 0(29026) check_self - checking if host==us: 9==9 && [10.0.0.11] == >>[10.0.0.11] >> 0(29026) lookup(): '0868594416' Not found in usrloc >> 0(29026) Warning: sl_send_reply: I won't send a reply for ACK!! >> 0(29026) receive_msg: cleaning up >> >> >>My Version of ser is: >># ser -V >>version: ser 0.8.11 (i386/freebsd) >>flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, >>SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>MAX_URI_SIZE 1024, BUF_SIZE 65535 >>@(#) $Id: main.c,v 1.162.2.5 2003/08/27 16:04:35 andrei Exp $ >>main.c compiled on 11:29:10 Sep 5 2003 with gcc 2.95 >> >>Does anyone know hwta may be going wrong here? I'm working through docs >>right now, but some guidance would be gratly appreciated; >> >> >>My vega 50 version details: >>Vega50(ISDN) Runtime System >>Version: 06.02.05.1 >>Built: Aug 19 2003 12:21:46 T018 >>Serial #:0050580106bd >> >>Bootstrap System >>Version: 1.09(0ws) >> >>ISDN Interface >>Version: Not known >> >>FLASH Partition Information: >>Partition 1: H.323 Firmware >> Version: 06.01.05.1 >> Built: Jul 17 2003 12:28:38 T014 >> >>Partition 2: SIP Firmware (ACTIVE) >> Version: 06.02.05.1 >> Built: Aug 19 2003 12:21:46 T018 >> >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> > >

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[Serusers] dropping call after ~15 sec "I won't send a reply for ACK!!"
by Jev 08 Sep '03

08 Sep '03

Hi, I'm setting up a vega 50 BRI with ser, and I can make in/out bound calls, quality is great! I have a problem, when I dial from a sip phone to a land line, my call gets set up, and works great, but after approx 15 seconds it drops the call. Below is what I get in the logs from ser. 0(29026) DEBUG: is_maxfwd_present: max_forwards header not found! 0(29026) DEBUG: add_param: tag=4063815580 0(29026) end of header reached, state=29 0(29026) parse_headers: flags=256 0(29026) find_first_route(): No Route headers found 0(29026) loose_route(): There is no Route HF 0(29026) check_self - checking if host==us: 9==9 && [10.0.0.11] == [10.0.0.11] 0(29026) lookup(): '0868594416' Not found in usrloc 0(29026) Warning: sl_send_reply: I won't send a reply for ACK!! 0(29026) receive_msg: cleaning up My Version of ser is: # ser -V version: ser 0.8.11 (i386/freebsd) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.162.2.5 2003/08/27 16:04:35 andrei Exp $ main.c compiled on 11:29:10 Sep 5 2003 with gcc 2.95 Does anyone know hwta may be going wrong here? I'm working through docs right now, but some guidance would be gratly appreciated; My vega 50 version details: Vega50(ISDN) Runtime System Version: 06.02.05.1 Built: Aug 19 2003 12:21:46 T018 Serial #:0050580106bd Bootstrap System Version: 1.09(0ws) ISDN Interface Version: Not known FLASH Partition Information: Partition 1: H.323 Firmware Version: 06.01.05.1 Built: Jul 17 2003 12:28:38 T014 Partition 2: SIP Firmware (ACTIVE) Version: 06.02.05.1 Built: Aug 19 2003 12:21:46 T018

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[Serusers] problem with textops.subst()
by Felix Schmid 08 Sep '03

08 Sep '03

Hello, I am having trouble with the textops' subst method. I am trying to rewrite the Contact header field for certain requests - and according to the debug output it seems to work (apart from the fact the the log output is scrambled): -- textops: subst_f: replacing at offset 341 [Contact: <sip:felix@192.168.0.66:5064;transport=udp>;methods="INVIT] with [Contact: <sip:felix@beluga.homeunix.org:5064;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, B]E, CANCEL, NOTIFY, ACK" --- However, if I dump out the whole request directly after calling subst(), using 'exec_msg("echo REQUEST: ; cat - ; echo");', I get this: --- REQUEST: REGISTER sip:beluga.homeunix.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.66:5064 CSeq: 7656 REGISTER To: "Felix Schmid" <sip:felix@beluga.homeunix.org> Expires: 900 From: "Felix Schmid" <sip:felix@beluga.homeunix.org> Call-ID: 742144068(a)192.168.0.66 Content-Length: 0 User-Agent: KPhone/3.11 Event: registration Allow-Events: presence Contact: <sip:felix@192.168.0.66:5064;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" --- Why isn't the request in a rewritten state after subst() returns? regards, felix -- PGP-Key: http://belugalounge.net/~felix/Felix%20Schmid.asc

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[Serusers] where is ser.cfg used on iptel.org server?
by Felix Schmid 07 Sep '03

07 Sep '03

Hello, I remember having read somewhere that the 'ser.cfg' file used at 'iptel.org' can be downloaded somewhere in order to be used as an example for many of the functionalities ser offers. Unfortunately I can't remember where I read this, nor did I find it on the ftp server... :( Is this config file still available for download? I am mainly asking because I am looking for an answer for the following phenomenon: I have SER running on my home network (on the gateway). When I try to send an IM to my account at iptel.org using kphone and I use my gateway SER as an outbound proxy, everything runs smoothly; I get a message back from iptel.org that the IM will be delivered to me as soon as I login the next time (what will not happen until I solved the NAT problem ;)). Now, when I try the same without using my gateway as an outbound proxy, I get the beloved message from iptel.org that it doesn't like my private Contact address. The only way the two requests differ is that the one using the outbound proxy uses a record-route I have attached the ngrep logfile, if anyone can explain me this behaviour I'd be glad. regards, felix -- PGP-Key: http://belugalounge.net/~felix/Felix%20Schmid.asc

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[Serusers] vega 50 BRI & ser
by Jev 07 Sep '03

07 Sep '03

Hello all, I should be taking delivery of a vega 50 BRI gateway unit at the end of this week. I plan to use this unit with ser, and a couple of cisco ata-186's. While I am waiting for the unit, I thought it might be a good idea to ask here if anyone has past experience using the vega 50 with ser? I know it's a pretty non-specific question, but it might help me get a headstart when I begin to set everything up :) Cheers! -Jev

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sr-users September 2003