sr-users July 2006

sr-users@lists.kamailio.org

206 participants
414 discussions

[Serusers] Newbie question: Registration problem

by VIDOVIC Mato

Hi folks! It is couple of days I am struggling with a problem to get SER up and running. I want to run it under SUSE 10.1 in the first instance simply without MySQL, RADIUS etc. - I am using the SER distribution 0.9.6. To my problem: actually, I can start SER and I get the information as follows: ------------- Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 169.254.158.130 [169.254.158.130]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 169.254.158.130 [169.254.158.130]:5060 Aliases: tcp: localhost:5060 udp: localhost:5060 -------------------------- The locally started client X-lite registers with SER without a problem (command ./serctl alias show confirms this). However, it is the only one who registers - all other (HW phone snom360 and other soft clients on other machine) send their REGISTER messages, but there is no answer from SER (and no ICMP messages from the machine, what is normal and OK!). In the ethereal trace it may be seen that SIP REGISTER messages are coming to the machine and there is no answer. I would appreciate any hint where I can look to see the problem. Best regards, Mato

18 years, 7 months

[Users] ACK for 487 'Request Terminated' fails

by Augustin, Andrew (external)

Hello, I am using OpenSER v1.0. I have a problem with the following SIP scenario with openser: - A-Party invites B-Party to a call session. While B-Party is 180 'Ringing', A-Party CANCELs the call session that it first initiated ACK messages for the 487 'Request Terminated' seems to fail to close the corresponding transaction as 487 continue to be sent multiple times until timeout. The message trace shown below is part of the message trace produced for the above scenario displaying message frames in sequence F89 to F137. OpenSER is running on all proxies. Initially everything works as expected: INVITEs, 100 'Trying's and 180 'Ringing' messages are routed correctly from A to B and from B to A via the intermediate proxies (this is not shown below). While still 'Ringing', A-Party CANCEL's the call (sends a CANCEL message) message: F107 in trace below. This also initally routes OK. But I expect the call flow for this scenario to stop at message: F130 once the 487 'Request Terminated' has routed its way back from the B-Party to the A-Party and a final 'ACK' is sent by the A-Party to the previous proxy: Proxy-1. But what actually occurs is that OpenSER running on Proxy-1 automatically resends a 487 'Request Terminated' at message F131 identical to the '487' it first sent at message F121. This in turn prompts further ACKs and 487s until it finally times out at message F186 (not shown below). I have also included part of the OpenSER message trace set at log level 9 which corresponds to ACK messages F130 and F132 received at proxy-A which do not appear to close the transaction of the preceding 487 'Request Terminated' messages - identical 487 'Request Terminate' messages are later resent from Proxy-1. Can you help me to: - identify and resolve the problem of multiple resends of 487 'Request Terminated' messages by Proxy-1. Best regards, Andrew Augustin ------- A-Party Proxy-1 Proxy-2 Proxy-3 B-Party 10.1.195.134 10.1.195.134 127.0.0.1:50 127.0.0.1:50 10.1.195.208 |>F89 INVITE>| | | | |<us 100 F90<| | | | | |>F91 INVITE (sdp)------->| | | |<important to us 100 F92<| | | | |<INVITE F93<| | | | |>F94 100 tr>| | | | |>F95 INVITE>| | | | |<us 100 F96<| | | |<-------(sdp) INVITE F97<| | | |>F98 100 trying -- your >| | | |>F99 INVITE (sdp)-------------------->| | |<-------------------- Trying 100 F100<| | |<------------------- Ringing 180 F101<| | |>F102 180 Ringing ------>| | | | |<g 180 F103<| | | | |>F104 180 R>| | | |<------ Ringing 180 F105<| | |<g 180 F106<| | | | |<g 180 F106<| | | | |>F107 CANCE>| | | | | |>F108 CANCEL ----------->| | |<g 200 F109<| | | | | | |<ANCEL F110<| | | |<---- canceling 200 F111<| | | | |>F112 CANCE>| | | | |>F113 200 c>| | | |<----------- CANCEL F114<| | | | |<g 200 F115<| | | |>F116 CANCEL ------------------------>| | |>F117 200 canceling ---->| | | |<-------- Request Terminated 487 F118<| | |<------------------------ OK 200 F119<| | |>F120 ACK --------------------------->| | |>F121 487 Request Termin>| | | |<-------------- ACK F122<| | | | |<d 487 F123<| | | | |>F124 ACK ->| | | | |>F125 487 R>| | | | |<- ACK F126<| | | |<est Terminated 487 F127<| | | |>F128 ACK -------------->| | |<d 487 F129<| | | | |>F130 ACK ->| | | | | |>F131 487 Request Termin>| | | |<-------------- ACK F132<| | |<d 487 F133<| | | | |>F134 ACK ->| | | | | |>F135 487 Request Termin>| | |<d 487 F136<| | | | | |<-------------- ACK F137<| | Part of OpenSER log file log-level 9 corresponding to ACK messages F130 and F132 from above trace:- 3(12429) DEBUG: add_to_tail_of_timer[4]: 0xb60e0090 3(12429) DEBUG: add_to_tail_of_timer[0]: 0xb60e00a0 3(12429) DEBUG:tm:relay_reply: sent buf=0x828b9c0: SIP/2.0 4..., shmem=0xb60e1530: SIP/2.0 4 3(12429) DEBUG: cleanup_uac_timers: RETR/FR timers reset 3(12429) DEBUG:destroy_avp_list: destroying list (nil) 3(12429) receive_msg: cleaning up 4(12430) SIP Request: 4(12430) method: <ACK> 4(12430) uri: <sip:testuser_b@generaltests.genericdomain> 4(12430) version: <SIP/2.0> 4(12430) parse_headers: flags=2 4(12430) Found param type 232, <branch> = <z9hG4bK1BCF7778>; state=16 4(12430) end of header reached, state=5 4(12430) parse_headers: Via found, flags=2 4(12430) parse_headers: this is the first via 4(12430) After parse_msg... 4(12430) preparing to run routing scripts... 4(12430) DEBUG : sl_filter_ACK: to late to be a local ACK! 4(12430) parse_headers: flags=100 4(12430) get_hdr_field: cseq <CSeq>: <640> <ACK> 4(12430) DEBUG: add_param: tag=1F5CE0A1 4(12430) DEBUG:parse_to:end of header reached, state=29 4(12430) DEBUG: get_hdr_field: <To> [83]; uri=[sip:testuser_b@generaltests.genericdomain] 4(12430) DEBUG: to body ["testuser_b" <sip:testuser_b@generaltests.genericdomain>] 4(12430) DEBUG: get_hdr_body : content_length=0 4(12430) found end of header 4(12430) DEBUG: is_maxfwd_present: max_forwards header not found! 4(12430) DEBUG: add_param: tag=1AE725BF 4(12430) DEBUG:parse_to:end of header reached, state=29 4(12430) parse_headers: flags=200 4(12430) find_first_route: No Route headers found 4(12430) loose_route: There is no Route HF 4(12430) DEBUG: t_check: msg id=12 global id=11 T start=0xffffffff 4(12430) parse_headers: flags=ffffffffffffffff 4(12430) parse_headers: flags=78 4(12430) t_lookup_request: start searching: hash=18675, isACK=1 4(12430) DEBUG: RFC3261 transaction matched, tid=1BCF7778 4(12430) DEBUG: t_lookup_request: transaction found (T=0xb60dffe0) 4(12430) DEBUG: t_check: msg id=12 global id=12 T end=0xb60dffe0 4(12430) parse_headers: flags=40 4(12430) Warning: sl_send_reply: I won't send a reply for ACK!! 4(12430) ERROR: Error while sending reply 4(12430) DEBUG:destroy_avp_list: destroying list (nil) 4(12430) receive_msg: cleaning up 10(12436) DEBUG: timer routine:1,tl=0xb60e1d44 next=0xb60e010c 10(12436) DEBUG: timer routine:1,tl=0xb60e010c next=(nil) 10(12436) DEBUG: timer routine:4,tl=0xb60e1cc8 next=0xb60e0090 10(12436) DEBUG: retransmission_handler : reply resending (t=0xb60e1c18, SIP/2.0 4 ... ) 10(12436) 3(12429) SIP Request: 3(12429) method: <ACK> 3(12429) uri: <sip:testuser_b@generaltests.genericdomain> DEBUG:tm:t_retransmit_reply: buf=0x642a80: SIP/2.0 4..., shmem=0xb60e3f70: SIP/2.0 4 10(12436) DEBUG: add_to_tail_of_timer[5]: 0xb60e1cc8 10(12436) DEBUG: retransmission_handler : done 10(12436) DEBUG: timer routine:4,tl=0xb60e0090 next=0xb60e5f34 10(12436) DEBUG: retransmission_handler : reply resending (t=0xb60dffe0, SIP/2.0 4 ... ) 3(12429) version: <SIP/2.0> 3(12429) parse_headers: flags=2 3(12429) Found param type 232, <branch> = <z9hG4bK3f84.a2d5c811.0>; state=16 3(12429) end of header reached, state=5 3(12429) parse_headers: Via found, flags=2 3(12429) parse_headers: this is the first via 3(12429) After parse_msg... 3(12429) preparing to run routing scripts... 3(12429) DEBUG : sl_filter_ACK: to late to be a local ACK! 10(12436) 3(12429) parse_headers: flags=100 3(12429) DEBUG: add_param: tag=1F5CE0A1 3(12429) DEBUG:parse_to:end of header reached, state=29 3(12429) DEBUG: get_hdr_field: <To> [83]; uri=[sip:testuser_b@generaltests.genericdomain] 3(12429) DEBUG:tm:t_retransmit_reply: buf=0x642a80: SIP/2.0 4..., shmem=0xb60e1530: SIP/2.0 4 DEBUG: to body ["testuser_b" <sip:testuser_b@generaltests.genericdomain>] 3(12429) get_hdr_field: cseq <CSeq>: <640> <ACK> 3(12429) DEBUG: get_hdr_body : content_length=0 3(12429) found end of header 3(12429) DEBUG: is_maxfwd_present: max_forwards header not found! 3(12429) DEBUG: add_param: tag=1AE725BF 3(12429) DEBUG:parse_to:end of header reached, state=29 3(12429) parse_headers: flags=200 3(12429) find_first_route: No Route headers found 3(12429) loose_route: There is no Route HF 3(12429) DEBUG: t_check: msg id=14 global id=13 T start=0xffffffff 3(12429) parse_headers: flags=ffffffffffffffff 3(12429) parse_headers: flags=78 3(12429) t_lookup_request: start searching: hash=18675, isACK=1 3(12429) DEBUG: RFC3261 transaction matched, tid=3f84.a2d5c811.0 3(12429) DEBUG: t_lookup_request: transaction found (T=0xb60e1c18) 3(12429) DEBUG: t_check: msg id=14 global id=14 T end=0xb60e1c18 3(12429) parse_headers: flags=40 10(12436) DEBUG: add_to_tail_of_timer[5]: 0xb60e0090 10(12436) DEBUG: retransmission_handler : done 10(12436) DEBUG: timer routine:4,tl=0xb60e5f34 next=0xb60e44a4 10(12436) DEBUG: timer routine:4,tl=0xb60e44a4 next=(nil)

18 years, 7 months

[Users] Bye problem

by raviprakash sunkara

Hi Users I'm Using the openser with Nat Bu using the RTP.. After invite method rtp is also established between the caller and callee ... Audio is clear.. But when I send Bye request to Callee , it not hung upping to callee it still estallishing the call... I think that .... 1) openSER server is not getting the request from callee or caller 2) OPenser is respones the BYE .. 3) Router(firewall) gateways is blocking the Bye request , to pass the request to openser, 4) Route behind the UA's isbloacking.... Here I creaking and hung upping ... Plz hep this... -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology Kondapur, Hi-tech city, Hyderabad. www.hyperion-tech.com +91-9985077535

18 years, 7 months

[Users] username prefixes

by Mark Kent

I'm looking for opinions from people who may have already done something like this: Suppose I've got some entries in the location table that I want the whole world to get to, and some locations that I want only authorized users to get to. Some of these locations have $resources$ attached, but even so they aren't hard-coded in my config, they register just like JoeUser on a SIP phone. I'm thinking of handling this with a prefix on the username, like auth.foobar vs foobar, like this: if (!uri=~"^sip:auth\.") { log(1,"---not auth destination\n"); if (lookup("location")) { append_hf("P-hint: non-auth usrloc\r\n"); route(1); } } # rest of stuff requires authorization if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); return; } consume_credentials(); <snip rest of INVITE handling> Is this naive, stupid, already handled in XYZ, or reasonable? Thanks, -mark

18 years, 7 months

[Users] VPS Hosting

by Abdul

Hi all, Could anyone suggest me any company who can host my openser/Asterisk on VPS. The price should be between $30 - $60. i fount lot of about about http://lylix.net/, does anyone have any experience about this company? Your suggession will be high appriciated. Khan --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs.Try it free.

18 years, 7 months

[Serusers] Number registered

by Teran Jesus

Hello I want to validate and get the 2 first digits of the telephone number registered, if i have a ip phone registered with a number like this 5556000000, when i recieve a INVITE from this number, i want to get 55 and validate the number, are there something to do this? Can you help me?

18 years, 7 months

[Serusers] Re: [Serweb-users] newbie: fifo issue on user login page

by Andrey Kouprianov

Hi, Add fifo_mode=766 to your ser.cfg and restart ser to make change permanent. If you want to make a temporary change (i.e. for a single time ser is started), just chmod 766 /tmp/ser_fifo Andrey. On 7/20/06, Sprechenzi English <sprechensenglish(a)yahoo.co.uk> wrote: > > > hi all! im stil new to this list. actually i just started configuring serweb > yesterday. and find it quite interesting though i lack enough knowledge in > troubleshooting it. i just would like to seek some assistance from this list > on my little problem: > > when logging into the user page, i can enter but it displays a red warning > or some sort of alert like this: > > sorry -- cannot open write fifo > > can someone help me to get rid of this? i've read the previous digests from > the list and i didnt seem to find the solution to my problem. by the way did > find the solution on the issue on missed calls! thanks to this list. i'm > really hoping someone kind enough will teach or just direct me to some > helpful link to solve my problem! thanks again and more power! > > sprechenzi > _______________________________________________ > Serweb-users mailing list > Serweb-users(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serweb-users > > >

18 years, 7 months

RE: [Serusers] Remote sip and rtp analyzing

by İlker Aktuna (Koç.net)

Hi, These are all very nice tools but none of them allow remote capturing of RTP. I am looking for something which would also capture RTP and let us analyze it. Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, July 19, 2006 9:52 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Remote sip and rtp analyzing Have you looked at these? http://www.pernau.at/kd/voip/bookmarks-sip-test.html g-) İlker Aktuna (Koç.net) wrote: Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, These are all very nice tools but none of them allow remote capturing of RTP. I am looking for something which would also capture RTP and let us analyze it. Thanks, ilker ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, July 19, 2006 9:52 AM To: İlker Aktuna (Koç.net) Cc: serusers(a)iptel.org Subject: Re: [Serusers] Remote sip and rtp analyzing Have you looked at these? http://www.pernau.at/kd/voip/bookmarks-sip-test.html g-) İlker Aktuna (Koç.net) wrote: Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker C "-//W3C//DTD HTML 4.0 Transitional//EN"> Hi, I still couldn't find a nice way of remote sip & rtp packet capturing and analyzing. Is there any tool to do this ? Do I need to use rcap protocol ? How can I use it on SER ? Thanks, ilker ________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) Sent: Friday, July 07, 2006 2:39 PM To: serusers(a)iptel.org Subject: [Serusers] Remote sip and rtp analyzing Hi, I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server. SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture. What is the best solution to analyze sip and rtp packets remotely from a SER server ? I also would like to have some statistical output, if possible. Thanks for your comments. ilker <http://387555.sigclick.mailinfo.com/sigclick/000A0602/040D4903/0004054B/032…> _____________________________________________________________________________________________________________________________________________ Bu e-posta mesaji kisiye ozel olup, gizli bilgiler iceriyor olabilir. Eger bu e-posta mesaji size yanlislikla ulasmissa, icerigini hic bir sekilde kullanmayiniz ve ekli dosyalari acmayiniz. Bu durumda lutfen e-posta mesajini kullaniciya hemen geri gonderiniz ve tum kopyalarini mesaj kutunuzdan siliniz. Bu e-posta mesaji, hic bir sekilde, herhangi bir amac icin cogaltilamaz, yayinlanamaz ve para karsiligi satilamaz. Bu e-posta mesaji viruslere karsi anti-virus sistemleri tarafindan taranmistir. Ancak yollayici, bu e-posta mesajinin - virus koruma sistemleri ile kontrol ediliyor olsa bile - virus icermedigini garanti etmez ve meydana gelebilecek zararlardan dogacak hicbir sorumlulugu kabul etmez. This message is intended solely for the use of the individual or entity to whom it is addressed , and may contain confidential information. If you are not the intended recipient of this message or you receive this mail in error, you should refrain from making any use of the contents and from opening any attachment. In that case, please notify the sender immediately and return the message to the sender, then, delete and destroy all copies. This e-mail message, can not be copied, published or sold for any reason. This e-mail message has been swept by anti-virus systems for the presence of computer viruses. In doing so, however, sender cannot warrant that virus or other forms of data corruption may not be present and do not take any responsibility in any occurrence. _____________________________________________________________________________________________________________________________________________

18 years, 7 months

[Serusers] SER/OpenSER + Asterisk Voicemail

by Marc Lewis

I've got OpenSER 1.0.1 working very well taking care of SIP UA's and have them going into asterisk voicemail correctly using Asterisk Realtime and MySQL 5.0 views. To ease our customers into this and give them a signle sign-on for all of the services we offer, we're using their username/email address on our main system. This gives them a single username/password combination for their POP email, DSL connection, webmail access, Jabber services and (soon) SIP services. My problem is this: Its no problem getting Asterisk to accept an alphanumeric mailbox for storing the user's voicemail. The problem is when they want to access it. If they're on a Linksys or Grandstream ATA and want to dial into their voicemail, it prompts them for their mailbox number. If their mailbox number is alphanumeric there is no way for them to enter that and they won't be able to get at their voicemail. What I would like to do is assign a mailbox number that is equal to their "primary" phone number. This can be stored in the usr_preferences table as an AVP. What I haven't figured out yet is how to rewrite that RURI before it sends it off to the asterisk server for voicemail. By doing this, we could use alphanumeric SIP logins, and then phone numbers for voicemail boxes. Given this configuration, how would I rewrite the user? I've tried: modparam("avpops", "avp_aliases", "mailboxuser=s:801") And then: avp_write("$ruri/username", "s:mailboxuser/g"); rewritehostport("10.0.0.2:5060"); append_branch(); But the user never changes. I've tried various things with rewriteuser and AVP's but haven't had any success so far. The usr_preferences row looks like this: +------+----------+--------+-----------+------+------------+ | uuid | username | domain | attribute | type | value | +------+----------+--------+-----------+------+------------+ | | sipuser | | 801 | 0 | 4255551212 | +------+----------+--------+-----------+------+------------+ Anyone have any suggestions on how to solve this problem? Thanks. - Marc -- Marc Lewis Avvanta Communications Corporation

18 years, 7 months

[Users] Parallel ringing

by Andreas M.

Hello, how could it be done through AVP´s and database lookup, to do a parallel ringing. Following setup: If uri eg example(a)test.dom is called, this call should be forwarded in parallel to a pstn-gateway. That means, receive the AVP with the telephonnumber associated to uri example(a)test.dom, rewrite uri, rewrite host/port and forward in parallel to gateway. I tried "append_branch", but was not able to put in the AVP (phonenumber), rewrite host/port points directly to the gateway. I have no troubles to load the AVP, only to make is possible, that the SIP Client and PSTN Phone are ringing in parallel. Some suggestions ? regards, Andreas

18 years, 7 months

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sr-users July 2006