sr-users April 2003

sr-users@lists.kamailio.org

69 participants
115 discussions

[Serusers] ser, pstn gateways
by Jev 27 Jun '03

27 Jun '03

Hello all, Im using ser, with esteras softphone, and a couple of cisco ata186's. I have to say I really like ser, as it is small fast, and reliable without being over complicated. I would like to know what other ser users and developers use for pstn gateways? I have a openline4 card from voictronix, but there is no software support for sip (When I bought it I thought I would use h.323). I have looked at cisco's pstn gateway modules, but was turned off them because of price, and lack of clear straight forward information to name but two reasons. Ideally what I would like to have is a solution that would support at least 4 pstn lines (expandable to maybe 8), that would run on FreeBSD (or at worst linux), works well with sip & ser, with high quality audio, and minimal delays when dialing in out. TIA, -Emil

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[Serusers] can't register
by Serge Kravchenko 12 May '03

12 May '03

hello. i've been trying to register to ser with couple of UAs like x10, estara and snom100. the response is following: RECEIVE << ser_ip_address:5060 SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP my_ip_address:5060 From: <sip:krav@sip.sipuser.com> To: <sip:krav@sip.sipuser.com>;tag=53d3b67acb7f1aa5eed7d0549c34647b.dea0 Call-ID: 77FCCA606A8111D7B7A2000021F0A5BA(a)sip.sipuser.com CSeq: 1788 REGISTER Server: Sip EXpress router (0.8.10 (i386/freebsd)) Content-Length: 0 Warning: 392 212.58.142.4:5060 "Noisy feedback tells: pid=93155 req_src_ip=ser_ip_address in_uri=sip:sip.sipuser.com out_uri=sip:sip.sipuser.com via_cnt==12" mysql has been installed properly, and i can add subscribers using serctl...... i guess there's something wrong with script, which is default, i've just uncommented everything regarding auth and mysql. and following is what i get in ser box logs: May 12 09:08:57 sip ./ser[93190]: ERROR: parse_cseq: bad cseq May 12 09:08:57 sip ./ser[93190]: ERROR: get_hdr_field: bad cseq May 12 09:08:57 sip ./ser[93190]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed any suggestions please? thanks.

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[Serusers] User Based Dialplans
by Fernando Nachtigall 07 May '03

07 May '03

Hi All, I need to implement an used based dialplans module on SER, so that depending on the user information stored in the database, a certain prefix should be inserted in the called number. Has anyone had a similar issue? Thanks Fernando --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.474 / Virus Database: 272 - Release Date: 4/18/2003

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[Serusers] using aliases
by Phil Yuska 30 Apr '03

30 Apr '03

Howdy, I am trying to determine why a call won't connect when dialed using an alias. I'm running SIP Express Router v0.8.10 on FreeBSD 4.8-RELEASE. Below is the message the client (Xten X-Lite) is receiving and apparently discarding. It appears as though the call is being routed but I'm puzzled by these message headers. RECEIVE << 192.168.1.11:5060 INVITE sip:pyuska@192.168.1.25:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKad62.cf628db7.0 Via: SIP/2.0/UDP 192.168.1.15:5060 From: yuska <sip:pyuska-ppc@shitworks.com>;tag=585911808 To: <sip:5315@shitworks.com> Contact: <sip:pyuska-ppc@192.168.1.15:5060> Call-ID: 06E1E114-22A1-8B9D-2891-72E17E9928EA(a)192.168.1.15 CSeq: 12625 INVITE Content-Type: application/sdp Content-Length: 283 Do I need to configure the SER server re-write the To: address or is the client misbehaving by not responding to the Contact: ? Any light you could shed on this would be appreciated. Phil

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Re: [Serusers] information about registered users
by Jiri Kuthan 30 Apr '03

30 Apr '03

At 12:14 PM 4/30/2003, radan(a)nasty.gda.pl wrote: >Hello all ! >I'm new user of ser :) > >I'd like have a information about registered users in my syslog > > save("location"); > log(3,"REGISTER zarejestrowany uzytkownik $USER"); > sl_send_reply("200", "ok"); ^^^^^^ > break; > }; > > >is some variable to use ? currently not. We are planning it for the future. -Jiri

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[Serusers] information about registered users
by radan＠nasty.gda.pl 30 Apr '03

30 Apr '03

Hello all ! I'm new user of ser :) I'd like have a information about registered users in my syslog save("location"); log(3,"REGISTER zarejestrowany uzytkownik $USER"); sl_send_reply("200", "ok"); ^^^^^^ break; }; is some variable to use ? Andrzej Radke sip:radan@task.gda.pl

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[Serusers] How to calculate new HASH_TABLE_SIZE value when extending parser?
by Maxim Sobolev 29 Apr '03

29 Apr '03

Hi, I am currently trying to extend ser's parser to allow it recognize User-Agent headers, but stuck into the magic HASH_TABLE_SIZE value. Is there any algorithm to calculate it when adding support for a new header type? Maybe there is some documentation, apart from comments in the code itself, on extending parser, or at least on its inner details? If so, it would be nice to obtain it if possible. Thanks! -Maxim

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[Serusers] Extending usrloc module to allow to save User-Agent values [patch for review]
by Maxim Sobolev 29 Apr '03

29 Apr '03

Folks, Attached please find a patch, which extends usrloc/registrar modules to save values from User-Agent field in REGISTER messages into the database. It would be nice to have it included into the next release. Please disregard hackish detection of User-Agent fields, I am currently working on extending parser to recognize that type of field. I'll post updated patch when finished. Also, there is a new unrelated function which can be used to retrieve full list of all currently registered users in all domains. I am using this function in new version of my nathelper modules to do periodical UDP "pinging" of all registered users to keep their NAT bindings alive. I would like to have it reviewed and included into the next release as well. Thanks! -Maxim

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Re: [Serusers] Load Balancing
by Jiri Kuthan 25 Apr '03

25 Apr '03

At 03:33 PM 4/22/2003, Alejandro Olchik wrote: >I need some help to implement load balance >when sending invites to an external sip >proxy server. > >I would like to be able to balance load >between to IPs (10.0.0.1 and 10.0.0.2) and >use the backup IP when the primary one >fails. > >Below is the routing code I have: > > > if (!lookup("location")) { > rewritehost("10.0.0.1"); > if (!t_relay_to("10.0.0.1","5060")) { > sl_send_reply("404", "Not Found"); > }; > break; > }; > >How can I add this behaviour? That depends on the distribution scheme you would like to use. For example, you can implement a distribution function that splits requests based on some downstream weights. e.g., modparam("distrib", "weights", # ... # rewrite host-part with one of values using the distribution # 20-30-50 distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) t_on_negative("1") t_relay_to(); # ... reply_route[1] { # look at destinations that failed, and update their probabilities # according to some strategy; for example, temporary 0% weight -- # note that this would take shared memory for keeping the weights # and a timer update_weights(); # if coupled to TM, it can look at previous attempts to eliminate # retrying to a previously failed destination distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) } -Jiri

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RE: [Serusers] problem with cisco 2600 to pstn
by Dan Austin 25 Apr '03

25 Apr '03

On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec. Dan -----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers(a)lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ------------------------------------------------------------------------ ---- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ------------------------------------------------------------------------ ---- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ------------------------------------------------------------------------ ---- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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sr-users April 2003