sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

[Serusers] Incoming pstn call
by Olivier AIME 12 May '04

12 May '04

Hi, When i implement, with a Cisco 3660, incoming PSTN call to serusers i have a problem. Call are well redirected to the right ip phone user, but when this user (here with XTEN or Cisco ATA) hangup, the soft phone user is disconnected, but not the pstn user at the other side. In fact, in XTen, i see the Cisco 3660 PSTN gateway ip address (instead of ser domain). So, when XTen send the BYE message to SER, the From uri look like HYPERLINK "mailto:user@"user@<CISCO3360PSTN>. PSTN user is not disconnected because this BYE message is "rejected" by ser. What did i miss ? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.674 / Virus Database: 436 - Release Date: 02/05/2004

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[Serusers] RE: [Serdev] call drop
by Helge Waastad 12 May '04

12 May '04

Hello, I don't know if anyone has written anythine else on this matter, but I agree that the ACK's usually are misrouted. In my case, I add a 0 (zero) before a forward the call to the cisco gateway. When SER receives an ACK to sip:0<number>, it is not caught by the "PSTN" route, but is caught by the "if(!lookup())" script, and hence (in my case) forwarded to handling_unavailiable_users (voicemail). Right now, as an interim solution, I've modified the !lookup to exclude ACK's, but I do not think this is the right way of doing it. My debug was like this: 0(23172) method: <ACK> 0(23172) uri: <sip:048095461@yyy.yyy.yyy.yyy:5060> 0(23172) version: <SIP/2.0> 0(23172) parse_headers: flags=1 0(23172) Found param type 235, <rport> = <n/a>; state=17 0(23172) end of header reached, state=5 0(23172) parse_headers: Via found, flags=1 0(23172) parse_headers: this is the first via 0(23172) After parse_msg... 0(23172) preparing to run routing scripts... 0(23172) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(23172) DEBUG : is_maxfwd_present: searching for max_forwards header 0(23172) parse_headers: flags=128 0(23172) get_hdr_field: cseq <CSeq>: <6909> <ACK> 0(23172) DEBUG: add_param: tag=61F29B28-E79 0(23172) end of header reached, state=29 0(23172) DEBUG: get_hdr_field: <To> [42]; uri=[sip:48095461@xxxx.xx] 0(23172) DEBUG: to body [<sip:48095461@xxxx.xx>] 0(23172) DEBUG: get_hdr_body : content_length=0 0(23172) found end of header 0(23172) DEBUG: is_maxfwd_present: max_forwards header not found! 0(23172) parse_headers: flags=64 0(23172) Someone's trying to register for Private IP, rewriting.... 0(23172) parse_headers: flags=64 0(23172) parse_headers: flags=33554432 0(23172) DEBUG: add_param: tag=547CFF73 0(23172) end of header reached, state=29 0(23172) parse_headers: flags=256 0(23172) check_self - checking if host==us: 10==12 && [yyy.yyyy.yyy.yyy] == [xxx.xxx.xxx.xxx] 0(23172) check_self - checking if port 5060 matches port 5060 0(23172) check_self - checking if host==us: 12==12 && [xxx.xxx.xxx.xxx] == [xxx.xxx.xxx.xxx] 0(23172) check_self - checking if port 5060 matches port 5060 0(23172) parse_headers: flags=256 0(23172) found end of header 0(23172) fnr(): No next Route HF found 0(23172) ras(): No next URI found 0(23172) lookup(): '048095461' Not found in usrloc 0(23172) query="select grp from grp where username='048095461' AND grp='voicemail'" 0(23172) is_user_in(): User is not in group 'voicemail' 0(23172) lookup(): '048095461' Not found in usrloc 0(23172) Enter voicemail route...... 0(23172) parse_headers: flags=44 0(23172) Warning: sl_send_reply: I won't send a reply for ACK!! br hw At 12:43 PM 4/8/2004, Raymond Chen wrote: >Then what should I do to take care of the problem? I'm not sure even what the problem is. If a call is dropped after one minute that's hardly related to mismatched reply. Similar behaviour often happens when an ACK is misrouted. Just watch SIP traffic to see what is going on. -jiri > I 've use the simple example and test every added step now. > >Ray > >-----Original Message----- >From: Jiri Kuthan [mailto:jiri <at> iptel.org] >Sent: Thursday, April 08, 2004 6:18 PM >To: Raymond Chen; serusers <at> iptel.org >Cc: serdev <at> iptel.org >Subject: Re: [Serdev] call drop > >At 03:17 AM 4/8/2004, Raymond Chen wrote: >>Dear all, >> >>WeÂ¡Â¯ve setup a call scenario AS5300-Â¨Â¤SERÂ¨Â¤as5300 but all calls drop around 1 minute after connection, > >SER does not keep transactions in memory for ever. If you wish them >to be kept longer, try longer TM timers. Nevertheless, the reply >in question should be delivered statelessly even if the transaction >context is already expired. > >-jiri > > > >_______________________________________________ >Serdev mailing list >Serdev <at> iptel.org >http://lists.iptel.org/mailman/listinfo/serdev -- Jiri Kuthan http://iptel.org/~jiri/

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[Serusers] pdt module
by Richard 12 May '04

12 May '04

Hi, I had a chance to try the pdt module. It works very well. In pdt module in source code, there is a very good README. Also under pdt/doc, there are several sql and php scripts. First create a database "pdt" and a table "domains". Grant ser access to the database if you want. The table domains has two columns "code" and "domain". For example, a domain abc.com has e164 number range 2345xyz. It uses xyz for internal extension and sip uri xyz(a)abc.com. You can manually add a record with 2345 as code and abc.com as domain. When a call comes in via pstn gateway to 2345678(a)1.2.3.4. The function prefix2domain translates it into 678(a)abc.com. What I don't understand is the terminator digit. It's a digit used by all domains to end a code. For example, if it is 5 and code is 234, prefix2domain translate 2345678 to 678 by removing code and then terminator. In North American, there is no terminator. Is there a way to bypass it? Another issue is that the function removes the code from original number. The assumption is that the domain owns the whole block. It is not always the case. For example, company A may have 2345200-2345399 and company B have 2345400-2345599. So code 2345 is possible to be mapped into two companies. We can't use two records mapping 23452 and 23453 to company A because it creates overlapped number. In another example, a company has two number blocks, 2345200-2345299 and 6789200-6789299. Two records can be defined in the database, map 23452 and 67892 to the same domain. Same problem, the rest of numbers are in an overlapped range. The problem can be solved if we can define how many digits are removed instead of removing all code digits. In the first example, we can define code 23454 and 23455 map to A.com, but only remove the initial four digits. So they rest of number are unique within company A. In the second example, we only remove the initial three digits, so 52xxx and 92xxx are still unique within the company. If this new number don't match the internal extension, we can always use alias to change it. This solution requires some change to the source code. Thanks, Richard __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

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[Serusers] How to check if callee is behind NAT ?
by jimmy huang 12 May '04

12 May '04

Hi all, now i can configure SER to check if caller is behind NAT , and modified SDP . forece using rtpproxy , so client can successed make voice call through NAT/Firewall but if caller is in public network , and callee is behind NAT , how to check if calee is behind NAT , and modify SDP , also force caller and callee to use rtpproxy in ser.cfg ?? now i can only check if caller is behind NAT or not but i don't find any related document has mention how to check if calee is behind nat , or in current version , NAT/Firewall traversal only works if caller is behind NAT ? can any one give me some idea ? thanks jimmy

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Re: [Serusers] fax over sip and ser
by Andreas Granig 12 May '04

12 May '04

Hi, please do also reply to the list. > Thanks, but how can I translate the tiff or something files to the fax > format? I thought you mean the SIP routing of fax calls on SER. So you mean how to send a fax directly from a computer? Therefore you'll need a SIP client capable of sending faxes via SIP. Is anybody here aware of such a client? I also searched for something like this without success some time ago. Another way would be to use a analog modem connected to a Cisco ATA or something similiar and send faxes via the provided fax software. Andy

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RE: [Serusers] Prepaid type access
by a.pletli＠euromacc.hu 12 May '04

12 May '04

As far as I know it's not possible with Vovida b2bua. I use b2bua just to session control with timeout... >Is it possible to have an ivr where it should tell the caller how many minutes do he/she have in his account using SER+Vovida b2bua + radius ? thanks.

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RE: [Serusers] Prepaid type access
by a.pletli＠euromacc.hu 12 May '04

12 May '04

Hi! Try SER+ Vovida b2bua + radius server for prepaid type access.You can find b2bua and documentation at www.vovida.org -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of wiggler Sent: Wednesday, May 12, 2004 4:46 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Prepaid type access Hi all, I would like to ask if its possible to have a sip server with a prepaid type of access? like allocating a time for a specific person. anyone can give me an idea or solution on how to do it if its possible? thanks a lot. //wiggler -- www.bembang.com

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[Serusers] how to connect calls to another sip server
by wiggler 12 May '04

12 May '04

Hi all, Im sorry to ask this question but its seems that im a newbie in sip and I would like to ask for help. I am going to create a sip server for every area in my place, area 1 username starts with number 1 and area 2 username starts with number 2, how can i forward my call from area 1 to area 2 is the user dials the number starting with number 2 and if the user dial a number starting with 1 it will go to local call only. i hope you can help me. thanks in advance. //wiggler -- www.bembang.com

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[Serusers] fax over sip and ser
by Andayesh 12 May '04

12 May '04

How can I implement fax over IP using SER? Any ideas? Thanks in advance...

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[Serusers] dbtext and serctl
by Rao 12 May '04

12 May '04

Is there a utility like serctl for dbtext ? dbtext works great but I would not want to setup the files by hand. Thanks. Rao. __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

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sr-users May 2004