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37456 discussions

[Users] Openser 1.0.1 installation problems
by J K 31 Jul '06

31 Jul '06

Hi there. I'm trying to install Openser 1.0.1 on CentOS 4.3. I've untarred the source code using 'tar zxvf openser-1.0.1_src.tar.gz' to a directory openser-1.0.1, but when I type 'make all' or 'make prefix=/ all' to build, I get the following errors: make: Warning: File `Makefile.rules' has modification time 1.4e+07 s in the future bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: Warning: File `Makefile.rules' has modification time 1.4e+07 s in the future bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: *** Deleting file `sr_module.d' make: *** [sr_module.d] Interrupt Anyone have any idea how I can fix this? Thanks in advance. __________________________________________ BreakThru.com - Pure email. Zero spam. http://friendfinder.com/go/f47930 - Dating Personals and Social Networking http://quizhappy.com/go/f301 - Ph.D certified tests and quizzes http://dine.com/go/f5554 - Restaurant Reviews in your area

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[Users] Resetting flags in branch_route[] resets flags for ALL branches
by Tavis P 31 Jul '06

31 Jul '06

I'm using branch route to trigger mediaproxy logic per branch for UAC directed calls and i'm experiencing some strange behavior: t_relay() is called, there are two branches In both branches there are two flags set During branch_route processing of the first branch both are left intact. During branch_route processing of the second branch, both are reset (unset) The destination UAC for the first branch picks up the phone call, sending a SIP OK message This is where i'm experiencing the problem Inside of the onreply_route both flags are NOT set, even though they were unset in a different call branch Has anyone experienced this behavior before? Is it supposed to work this way? This is using a recent (1-2 weeks ago) cvs checkout of OpenSER 1.0.0 tavis

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[Serusers] How to handle redirect with sipbroker?
by Roger Lewau 30 Jul '06

30 Jul '06

I dont know if any of you use the SIPBroker service to do multi enum lookups, but I'm hoping there are some on this list who do. I have been using SIPBroker for dialing with sip codes only so far and did my enum lookups on my own. But now, I decided to set ser up to try sipbroker first for all external calls and then fall back to my different gateways if no route can be found via SIPBroker. SIPBroker, in prinicpal, works like this. I send an Invite for any enum number to SIPBroker. If sipbroker finds a route for that number to another voipprovider it will proxy the call to the found provider. If sipbroker can not find a route, it will reply with a redirect of the call to my self and I am supposed to handle the call setup my self, through my gateways. My question is how to handle this redirect message? Any one who has a working failure route to handle this situation, and are willing to share? Here is an actual SIP conversation, initiated from Asterisk via SER, of a failed Enum lookup through SIPBroker: # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 INVITE sip:4640240252@sipbroker.com SIP/2.0. Record-Route: <sip:212.247.91.XXX;ftag=as5e811304;lr=on>. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>. Contact: <sip:330000@212.247.91.XXZ>. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sat, 29 Jul 2006 22:42:23 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 336. . v=0. o=root 850 851 IN IP4 212.247.91.XXZ. s=session. c=IN IP4 212.247.91.XXZ. t=0 0. m=audio 34852 RTP/AVP 0 8 18 97 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:97 iLBC/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. ## U 24.196.79.163:5060 -> 212.247.91.XXX:5060 SIP/2.0 300 Redirect. Via: SIP/2.0/UDP 212.247.91.XXX;branch=z9hG4bK7307.c7d88007.0. Via: SIP/2.0/UDP 212.247.91.XXZ:5060;branch=z9hG4bK05f7cabc;rport=5060. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. CSeq: 103 INVITE. Contact: sip:4640240252@sip.serverhallen.com. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 24.196.79.163:5060 "Noisy feedback tells: pid=15326 req_src_ip=212.247.91.XXXreq_src_port=5060 in_uri=sip:4640240252@sipbroker.com out_uri=sip:4640240252@sip.serverhallen.com via_cnt==2". . # U 212.247.91.XXX:5060 -> 24.196.79.163:5060 ACK sip:4640240252@sipbroker.com SIP/2.0. Via: SIP/2.0/UDP 212.247.91.237;branch=z9hG4bK7307.c7d88007.0. From: "Roger Lewau" <sip:330000@sip.serverhallen.com>;tag=as5e811304. Call-ID: 6877d50b1343fac25403e06f4a83c280(a)sip.serverhallen.com. To: <sip:240252@sip.serverhallen.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe4c. CSeq: 103 ACK. User-Agent: Sip EXpress router(0.9.3 (i386/freebsd)). Content-Length: 0. .

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[Serusers] Mediaproxy accounting - SER
by Robert Zorop 29 Jul '06

29 Jul '06

Hi, is there anyone experimenting problems with dbaccounting with mediaproxy ?. I have installed 1.7.2 and it works in some situations and doesnt in others. The calls are terminating in a cisco 2600 equip. The log i have is : delete 347395089(a)192.168.0.102 info= session 347395089(a)192.168.0.102: 93/3917/4010 packets, 17736/782484/800220 bytes (caller/called/relayed) session 347395089(a)192.168.0.102: ended. But no account is done. Sincerely i don't really know what is happening here. Please any help/clue provided will be a great help .

1 0

[Serusers] Implementing Group Pickup
by Simon 29 Jul '06

29 Jul '06

Hi, Has anyone thought how to implement group pickup purely in SER ? Currently we are using Asterisk for all PBX type functions, but want to move to a design where SER handles the dialogue with the sip phones and uses Asterisk purely as a voice mail server. So has anyone implemented group pickup and hunt groups before ? Many Thanks Simon

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[Serusers] Problem compiling SER with Radius
by Marco Dias 29 Jul '06

29 Jul '06

I installed SER and everything was working ok but I had to recompile SER with RADIUS support. I installed radiusclient 0.3.2 from sources. I editted the Makefile deleting "auth_radius", "group_radius" and "uri_radius" from the excluded modules. I Tryed to recompile SER but when I did "make install" appeared the following warnings: ERROR: module modules/acc/acc.so not compiled ERROR: module modules/auth_radius/auth_radius.so not compiled ERROR: module modules/group_radius/group_radius.so not compiled ERROR: module modules/uri_radius/uri_radius.so not compiled I tryed to locate the modules but only found the acc.so in "/usr/ local/lib/ser/modules/acc.so". The others I don't have... Can anyone help me please... Thanks in advance and sorry about my bad English Marco Passarinho

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[Serusers] Replacing CID to Alias
by Ali Zaidi 29 Jul '06

29 Jul '06

Hi, Is this possible to replace CID (SER UA ID) to Alias. I have UA 5001112222(a)mydoamin.com and Alias 2123334444. When UA call other UA2 the ID shows 5001112222, but i want to show 2123334444. Thanks, Ali...

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RE: [Serusers] Remote sip and rtp analyzing
by İlker Aktuna (Koç.net) 29 Jul '06

29 Jul '06

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/03010505/04074D00/030C044C/031…> _____________________________________________________________________________________________________________________________________________ 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. _____________________________________________________________________________________________________________________________________________

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[Serusers] Accounting tools or ideas needed
by Steve Blair 29 Jul '06

29 Jul '06

We are generating accounting records for INVITE, BYE and CANCEL methods. We use flatstore and rotate the files nightly via cron.daily. We expected the number of each type of record to "add up". That is we expected each INVITE to have either a BYE or CANCEL based on the Call-ID field but we are actually getting duplicates and in some cases seem to get BYEs without a corresponding INVITE. I think some of these BYE may be for calls initiated by the proxy to either our Asterisk voicemail server or to another UA because of blind transfers initiated at the phone or call forward always initiated by the proxy. The trouble is we have 600+ subscribers and the logs are huge. I have used cat, grep and sort to try and identify the mismatch but this is tedious at best. Does anyone have any insight into why the mismatch might exist and suggestions for record matching based on the Call-ID field. We are using ser 0.9.7-pre3, Cisco 79x0 phones running 7.3 image and Cisco 2821 gateways running 12.3(8)T11. Thanks,Steve

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[Serusers] altering the caller id (CID)
by Mark Price 29 Jul '06

29 Jul '06

Hi, Is there a common method for rewriting the caller id for outboud calls? apparently rpid is insufficient.. I need something like the "set(CALLERID=904xxxxxxx)" thing in asterisk. Thanks Mark price

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