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37537 discussions

[Users] Adding headers to the CANCEL messages generated by openser
by Ardjan Zwartjes 18 Apr '07

18 Apr '07

Hello openser users, I'm trying to do something with openser but I haven't been able to figure out how to do this. So I hope somebody on the list knows the answer: In my configuration script I use append_hf to add application data to the SIP messages, this works great until the moment where I do a parallel branch. The invites work just fine since I can do the append_hf in the branch_route. The problem arises when one of the branches is answered, at this point OpenSER will generate CANCEL messages for the other branches and I have no clue on how to add header fields with information about the original invite to those messages. So my question is: Does somebody know how you can add headers with data to the CANCEL messages generated by SER in a parallel branch? Thanks in advance. Kind regards, Ardjan Zwartjes.

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[Users] Diferences between "fix_contact" and "fix_nated_contact"
by Edson 18 Apr '07

18 Apr '07

Hi, Despite the fact that this two functions are declared on NATHelper and Mediaproxy modules, from a practice point of view, they appear to do the same, so when to apply one and another? Aren't there functionalities overlaped? Edson. ------ "It wouldn't be sell if it were easy..." (myself)

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[Users] prefix in dispatcher
by Jayesh Nambiar 18 Apr '07

18 Apr '07

Hi, I am using dispatcher module in my openser 1.2. I am using ds_select_domain function in the script. What I wanted to know is, is there a way that I can prefix some digits to the ruri before sending to the selected domain. My scenario is: I have got two Gateway IPs in the dispatcher module. Each Gateway needs a different prefix before the ruri to process the calls properly. I am using the round robin algo in dispatcher. Now the question is how will I know what IP is being selected and how will I attach appropriate prefix? Is there a way to do the same in the dispatcher module. My configuration snippet: route[4] { #--Dispatcher route-- ds_select_domain("1", "4"); $avp(i:25) = 3; $avp(i:30) = $rd; route(1); exit; } thanks in advance. w/regards, Jayesh Send a FREE SMS to your friend's mobile from Yahoo! Messenger. Get it now at http://in.messenger.yahoo.com/

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[Users] avp_check() replacement? Openser 1.2?
by Brandon Armstead 18 Apr '07

18 Apr '07

Hello, I'm wondering if in OpenSER 1.2 if there is a replacement method to check if a value is inside an avp list, as follows: if($avp(some_avp_value) == "$avp(some_avp_list_value)/g"){ # do something } avp check equivlent: if(avp_check("$avp(some_avp_value)", "eq/$avp(some_avp_list_value)/g"){ # do something } Does anyone know if this is implemented or being planned for implementation, what are everyone else's thoughts on something like this? Thanks! -- Brandon Armstead

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[Users] OpenSER status meeting 20-04-2007 13:00 GMT
by Henning Westerholt 18 Apr '07

18 Apr '07

Hello all, the first IRC OpenSER status meeting will take place this friday, the 20-04-2007 at 13:00 GMT (15:00 CEST) at the #openser channel on irc.freenode.net. The purpose of this meeting is to discuss the latest developments in and around the OpenSER project. Topics so far: 1. Experiences with OpenSER 1.2 2. OpenSER project server hosting 3. State of OpenSER Documentation 4. Next release For information about the IRC meetings read: http://openser.org/dokuwiki/doku.php/development:irc-meetings For details about this meeting read: http://openser.org/dokuwiki/doku.php/development:irc-meeting-agenda-20-04-2… Please join us! -- Henning Westerholt

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[Users] extraneous 408's
by Tim 18 Apr '07

18 Apr '07

Hello, Since upgrading to 1.2 we have begun to see lots of extraneous 408 messages. The CSEQ indicates they are associated with INVITE transactions. Weird thing is they happen well after the INVITE transaction is complete. Any ideas on how I can figure out what is going on here? These are generated by openser not proxied from another source. Tim

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Re: [Serusers] interesting 2 way SIP signalling problem
by Ladislav Andel 18 Apr '07

18 Apr '07

Hi, Looks like you are mixing together nathelper and mediaproxy module. You need to call force_rport() even for INVITEs. Look at Getting started ser configs ftp://siprouter.onsip.org/pub/gettingstarted/configs/ Lada İlker Aktuna (Koç.net) wrote: > Hi, > > Did you receive my reply yesterday ? > > Thanks, > ilker > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of İlker Aktuna (Koç.net) > Sent: Monday, April 09, 2007 11:29 AM > To: Ladislav Andel > Cc: serusers(a)iptel.org > Subject: RE: [Serusers] interesting 2 way SIP signalling problem > > Hi, > > This is under my method=="REGISTER" section: > > if (nat_uac_test("1")) { > setflag(6); > fix_nated_contact(); > force_rport(); > }; > > > I also have this under method=="INVITE" : > > if (method=="INVITE" || method=="ACK") { > > if (client_nat_test("19")) { > fix_nated_contact(); > }; > > And this one under a onreply route : > > onreply_route[2] { > if (status=~"(183)|(2[0-9][0-9])") { > use_media_proxy(); > > if (nat_uac_test("19")) { > fix_nated_contact(); > }; > } > }; > > > Thanks for your reply... > > ilker > > -----Original Message----- > From: Ladislav Andel [mailto:ladaan@iptel.org] > Sent: Monday, April 09, 2007 11:31 AM > To: İlker Aktuna (Koç.net) > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] interesting 2 way SIP signalling problem > > Hi, > > do you call force_rport() function for NATed clients? > Can you paste in the NAT section of your ser.cfg here? > Paste in NAT detecting section too. > > Lada > > > İlker Aktuna (Koç.net) wrote: > >> Hi, >> >> I have a working configuration on my SER server. Most of the users can make calls (both way incoming and outgoing) with no problem. >> However, I have some clients that can't make calls. >> When I analyze these by ngrep and/or tcpdump, I see that the SER server does not reply the packet on the correct port. >> >> For example, I have a testing client which is behind NAT. >> The client sends INVITE message from UDP port 3023 to port 5061 of my SER. >> (cseq number is 1) >> SER exchanges the message with the uplink sip gateway and when it >> receives a "OK" response it forwards this message to the client's NAT >> IP address BUT on a different port (14640) with correct cseq number >> (1) >> >> Of course, this new port is not open on the NAT router and it does not reach the client. >> >> I see that the first INVITE message has Contact field with internal IP address of the client and the port number as 14640. >> So the SER server sends the returning message to the external IP address with internal port number. >> >> How can I fix this issue ? >> >> I can send the configuration of my SER server if that would help to find the problem. >> > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _____________________________________________________________________________________________________________________________________________ > 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] how-to "loop" in the ser.cfg file
by Steve Blair 17 Apr '07

17 Apr '07

Hello: I need to perform a check several times within a failure_route and I'm not sure how to accomplish this task. The way I implemented sequentially hunting I perform the relay to the first number in the hunt list within route block 0. If this number is unavailable I "fail" to a failure block then interate through the remaining numbers in the huntlist in that failure route but this presents a problem. All of the "feature" and "call permission" checks which I normally perform on an extension are only performed in route block 0. This means I am not checking the feature and call permission flags for the 2nd through Nth number in the huntlist once I enter the failure route. I created a route block to perform the necessary checks but the problem is I do not see how to return to the failure route if the feature check for a given number fails. For example. If the t_relay to extension 1234 fails I will enter failure_route #1. In failure_route #1 I load the next extension, 2345, but if the call permission check for 2345 fails how do I skip this number in the huntlist, load the next number in the list and repeat the failure_route #1 processing? Thanks,Steve

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[Users] SEMS: ivr: mp3 module
by Helmut Kuper 17 Apr '07

17 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have sems with voicemail module and mp3 support running. I modified it to save recorded mp3 files on harddisc. I modified ivr's mailbox_query module to load that prevoiusly on harddisc saved mp3 files. Unfortunately the male mp3 library isn#t able to decode/read/play that mp3 files. Is there any change that there will be mp3 decoding support in mp3 module? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGJOTm4tZeNddg3dwRAhxdAJ9UjRKha1hqV52MN70uwagyLMawVACfQwz8 FnrVEKj0ln45hZmlA9S2xYk= =2q7+ -----END PGP SIGNATURE-----

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[Serusers] question re SER capacity and redirecting using rURI contents.
by Tom Meadows 17 Apr '07

17 Apr '07

hi- I am looking into using SER as part of our test infrastructure for a SIP-based messaging system and have 2 questions: one about SER capacity, and one about the best way to configure it. Firstly though, the setup: We use a script-driven inhouse SIP/RTP test tool in association with our messaging system which can either send or accept SIP invitations and process the associated RTP streams (eg act as UAS or UAC). This tool has a capacity of ~125 ports. We need the messaging system to create around 800 concurrent SIP/RTP sessions (dial out), and our system allows only one outbound sip gateway. I therefore need to configure a sip redirector which can take incoming SIP invites from the messaging system and redirect them to one of 7 test tool clients (125x7 > 800). First question, re capacity - the 800 concurrent sessions will be staggered and spread over 2 minutes. This works out at about 7 SIP invites per second. Should SER be able to handle this (proposed SER box =2.8Ghz, 512mB RAM). Secondly, the Request-URIs of the SIP invites sent by the messaging system are structured, and it is easy to tell from the Request-URI which client machine we want this invite to be redirected to. Eg: 1000000000-1000000999(a)dom.com -> should go to client 1 1000001000-1000001999(a)dom.com -> should go to client 2 1000002000-1000002999(a)dom.com -> should go to client 3 1000003000-1000003999(a)dom.com -> should go to client 4 etc Rather than registering all these accounts on SER, is there an alternative way I can get SER to redirect various SIP invites using wildcards to check the structure of their r-URI? eg 1000000...(a)dom.com redirects to A 1000001...(a)dom.com redirects to B 1000001...(a)dom.com redirects to C etc. Thanks for advice. Tom _________________________________________________________________ Txt a lot? Get Messenger FREE on your mobile. https://livemessenger.mobile.uk.msn.com/

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