sr-users

sr-users@lists.kamailio.org

4 participants
37456 discussions

RE: [Serusers] decoding g729 encoded rtp streams
by İlker Aktuna (Koç.net) 26 Jul '06

26 Jul '06

Hi, Thanks for your kind reply. Unfortunately I couldn't find any implementation, but in the thread you submitted, seems that Mr. Martin Regner has made an application to do what I want to achieve. Is there a way to enter that forum to contact him ? http://www.ethereal.com/lists/ethereal-dev/200309/msg00152.html Thanks, ilker -----Original Message----- From: Andres [mailto:andres@telesip.net] Sent: Tuesday, July 25, 2006 5:52 PM To: İlker Aktuna (Koç.net); serusers(a)iptel.org Subject: Re: [Serusers] decoding g729 encoded rtp streams İlker Aktuna (Koç.net) wrote: >Hi, > >Thanks for the detailed explanation. It's really useful. >I'm trying to automate this whole process. All tools except Ethereal (tcpdump, rtpdump, voiceage decoder) are command line tools that can be used in a batch/shell script. >How can I export rtp stream without using Ethereal, or from command line ? > >is there a way ? > > There probably is a way but I have not have the need to investigate. You can try googling around and you will find related discussions like: http://www.ethereal.com/lists/ethereal-dev/200309/msg00152.html >Thanks, >ilker > > >-----Original Message----- >From: Andres [mailto:andres@telesip.net] >Sent: Monday, July 24, 2006 8:18 PM >To: İlker Aktuna (Koç.net) >Cc: serusers(a)iptel.org >Subject: Re: [Serusers] decoding g729 encoded rtp streams > >İlker Aktuna (Koç.net) wrote: > > > >>Hi, >> >>Thanks for the information. >>Actually, I downloaded it but couldn't find how to export rtp payload. >>Does rtpdump read capture files recorded with Ethereal/tcpdump ? >>(in which format should I save the file?) >> >>Btw, there is a Windows sniffer named "Cain & Abel" this one decodes g729. >> >>Thanks, >>ilker >> >> >> >First you export the RTP data from within Ethereal like this: >1. Open Capture >2. Click on Analyze->Statistics->RTP Streams->Show All 3. Select the stream you want to export 4. Click 'Save As' >5. Then feed that file into rtpdump like the below example 'trace.rtp'. > >Here is the command line: >rtpdump -F payload -f trace.rtp -o file.out > >file.out will contain just the payload which can then be decoded. > > > > >>-----Original Message----- >>From: Andres [mailto:andres@telesip.net] >>Sent: Friday, July 21, 2006 10:53 PM >>To: İlker Aktuna (Koç.net) >>Cc: serusers(a)iptel.org >>Subject: Re: [Serusers] decoding g729 encoded rtp streams >> >>İlker Aktuna (Koç.net) wrote: >> >> >> >>>Hi, >>> >>>I know that there are licence and patent issues but is it possible to >>>decode g729 encoded rtp streams with any software ? >>>I have licenced Eybeam softphone with g729 support. But it can only >>>decode streams in a conversation. >>>I just need to decode streams recorded with tcpdump/ethereal. >>> >>>Once I had used a software with this feature but then it turned out >>>to be a trojan. I don't remember the name now. >>>Is there any other software ? (or method using Eyebeam ?) >>> >>>Thanks, >>>ilker >>> >>> _____________________________________________________________________________________________________________________________________________ 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. _____________________________________________________________________________________________________________________________________________

2 1

[Serusers] DNS SRV Mediaproxy
by Richard C. Thompson 26 Jul '06

26 Jul '06

Hi Can anyone post a hint so I can figure this out. I have a ser server using a DNS lookup for my SRV mediaproxy servers. Everything works fine, and has been for some time. Now I built a new ser server and want to go to the same mediaproxy servers but I get the following error Jul 26 09:47:36 localhost proxydispatcher[31666]: warning: none of the proxies for domain mydomain.com have responded Jul 26 09:47:36 localhost proxydispatcher[31666]: error: No mediaproxy servers found for the default domain. I have included the new ser IP in the allow config of the mediaproxy Even if I force the defaultproxy to one of the IP addresses of a mediaproxy server I get the following error error: couldn't forward command to 0.0.0.0:25060 (Connection reset by peer) Any ideas??? Could it be a version issue or maybe the python-mysqld issue, I'm not using accounting. Rick

2 1

[Serusers] ser 0.9.7-pre3 SRV "question"
by Steve Blair 26 Jul '06

26 Jul '06

Hello: We are running ser 0.9.7-pre3 and seeing some strange behavior. I'm hoping someone on this list might have some input. Our SIP domain name is actually an SRV service name that points to two A records. This configuration has existed for years and has worked reliably. _sip._udp.myserver.myschool.edu _sip._udp.myserver.myschool.edu service = 1 5 5060 proxy1.myschool.edu. _sip._udp.myserver.myschool.edu service = 3 5 5060 proxy1.myschool.edu. The "myserver.myschool.edu" name is not the real name it is just for discussion purposes. The name "myserver.myschool.edu" has other SRV records associated with it. For example: $nslookup > set type=SRV >_kerberos._udp.myserver.myschool.edu _kerberos._udp.net.isc.upenn.edu service = 0 100 88 trixie.myserver.myschool.edu _kerberos._udp.net.isc.upenn.edu service = 0 100 88 pops.myserver.myschool.edu In the past month we upgraded to this release of SER and noticed a problem when users call forward their extension to an off campus PSTN number. If the Call Forward Always flag is set to Y then the R-URI is re-written to be a new user portion with our SIP domain name appended as the hostname. In some ngrep traces I'm seeing our proxy trying to send the forwarded call to a next hop address of "pops.myserver.myschool.edu" as shown above. The problem is pops is not a SIP server. The only connection between our SIP environment and the host pops is the domain name. This suggests an issue with SRV resolution but I do not see the problem in my traces. Does anyone have any thoughts on this issue? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

2 9

Re: [Users] Ringing timeout??
by Zenone 26 Jul '06

26 Jul '06

--------- Message d'origine -------- De: Klaus Darilion <klaus.mailinglists(a)pernau.at> A: Zenone <zenone(a)telip.com> Copies: users(a)openser.org Objet: Re: [Users] Ringing timeout?? Date: 26/07/06 15:37 > >> > >> > > Thanks! But is there a way to set a particular fr_inv_timer for a call? > > Michel > > > Yes, via AVP. Read the tm docs :-) > > Thanks a lot, Michel ________________________________________________ Message sent using UebiMiau 2.7.8

1 0

Re: [Users] Ringing timeout??
by Zenone 26 Jul '06

26 Jul '06

--------- Message d'origine -------- De: Klaus Darilion <klaus.mailinglists(a)pernau.at> A: Zenone <zenone(a)telip.com> Copies: users(a)openser.org Objet: Re: [Users] Ringing timeout?? Date: 26/07/06 14:52 > Zenone wrote: > > Hi! > > Is it possible to limit the ringing duration? For example to give up before > > going to the called number voicemail... > > fr_timer and fr_inv_timer parameter of tm module (refer to the tm module > docs) > > regards > klaus > > > > Thanks in advance, > > Michel > > > > ________________________________________________ > > Message sent using UebiMiau 2.7.8 > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > Thanks! But is there a way to set a particular fr_inv_timer for a call? Michel ________________________________________________ Message sent using UebiMiau 2.7.8

3 2

[Users] Ringing timeout??
by Zenone 26 Jul '06

26 Jul '06

Hi! Is it possible to limit the ringing duration? For example to give up before going to the called number voicemail... Thanks in advance, Michel ________________________________________________ Message sent using UebiMiau 2.7.8

2 1

[Users] TCP problems under high load
by Klaus Darilion 26 Jul '06

26 Jul '06

Hi! I suspect openser to have TCP problems under high load: Scenario: sipp sends INVITE to openser, which replies with 404 I have 4x sipp with 1000 INVITEs/s and 1x sipp with 3000INVITEs/s, thus 5 TCP connections After some time, I get error on the TCP connection with 3000INVITEs/s: -->syslog.txt Looks like the first part of the request is missing, but sniffing revealed that the request is proper formated as you see in the TCP packet -->ngrep.txt Are there known high-load problems? Any ideas how I can debug this? regards klaus

2 3

[Users] Dial Plan to Redirect To Message
by openser＠comcast.net 26 Jul '06

26 Jul '06

Anyone know how to use dial plan to play messages as soon as a phone is picked up. Like when a user picks up a phone, get a message to contact administrator instead of a dial tone?

3 2

[Serusers] Ringing timeout??
by Zenone 26 Jul '06

26 Jul '06

Hi, I would like to know if it's possible to set a ringing timeout for particular calls?And if it exists, what is the way to do this? Thanks in advance for your help, Michel ________________________________________________ Message sent using UebiMiau 2.7.8

2 1

[Users] Some trouble with multiple failure routes.
by Brandon Price 26 Jul '06

26 Jul '06

I am having trouble setting up a failure route to roll over to another on failure. I have failure_route[1] set up to catch 302 redirects and act accordingly. If no redirect is detected then the INVITE should roll to failure route 2, as per the line that says t_on_failure("2");. This is however not what occurs. When an INVITE comes in, and hits failure_route[1] I can see in /var/log/messages that it hit that route, however it never gets to failure_route[2] even though there is no 302, instead the 408 is sent to my sip provider, they ack it, and the call is dropped. Any help is as always greatly appreciated. ######################################################################## ################################# # FAILURE ROUTE 1 --- 302 REDIRECT --- ######################################################################## ################################ failure_route[1] { xlog("got to failure route 1"); t_on_failure("2"); if(t_check_status("302")){ xlog("$tU has a forward"); get_redirects("*", "redirect"); t_relay("my.sip.proxy"); exit; }; } ######################################################################## ############################### # FAILURE ROUTE 2 ----- VOICEMAIL ------- ######################################################################## ################################ failure_route[2] { xlog("got to failure route 2"); xlog("$tU getting a vmail"); if (t_check_status("408|486|487")){ revert_uri(); strip(1); prefix("201"); rewriteport("5070"); append_branch(); t_relay(); exit; }; }

2 2

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users