sr-users January 2006

sr-users@lists.kamailio.org

134 participants
258 discussions

[Users] How to check Displayname in From Tag

by martin＠campus-merseburg.de

Hi, i want to change the "From:" tag with uac_replace_from when the displayname is "anonymous". with from_uri i can only check the uri. is there a way to check the displayname in the "From: " tag? for example: From: "Anonymous <sip:1234@blah.net>" i want to change it to: From: "Anonymous <sip:Anonymous@Anonymous>" Ciao

18 years, 9 months

[Serusers] SER RTP QoS Monitoring

by Aisling O'Driscoll

Hi, I want to be able to extract QoS parameters such as jitter, delay and packet loss from VoIP calls made using my SER(both peer to peer for my public clients and also for clients where the voice will be routed with mediaproxy). I would like to develop something as opposed to purchasing a software package. >From searching the Internet the last few days I have seen that the following options seem to be available: 1) RTCP RTCP stream will contain this information. The problem is that sniffers cannot clearly distinguish RTCP packets from other UDP packets unless something like an IP address or specific UDP port is supplied. I had considered using TCPDUMP but then where do I position this in the network when voice is going peer to peer between client and I'm not sure how this can work to monitor all calls when I might not be aware of the end user IP addresses. 2) SIP INFO extension This also seem to be an option but most off the shelf phones don't seem to support this and this would required modification of a SIP ua. 3) SNMP I thought maybe SNMP might be an option but the SER snmp module no longer exists... Does anyone have any comments on the above? Are the statements that I made correct or can anyone think of other ways to monitor the voice QoS? I am trying to understand how commercial applications have accomplished this. Many thanks in advance for advice. Aisling. >---- Original Message ---- >From: clona(a)cyberhouse.no >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Status of SER SNMP module >Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> >>Hi Aisling, >> >>The SNMP module is dead as a killed turtle :( >> >>-Atle >>* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]: >>> Hello, >>> >>> I'm just wondering what the current status of the SER SNMP module >is? >>> Is it currently supported and is there somewhere I can find >>> documentation on it? >>> >>> Many thanks, >>> Aisling. >>> >>> >>> >>> -------------------Legal >Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and >intended only for the person to whom it is addressed. Its contents >may be protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)iptel.org >>> http://mail.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

18 years, 9 months

[Serusers] Re: 200 OK/SDP and ACK/SDP problem

by Carla Ares

> > Hi all, > I have a problem with audio, when the destination of a call is natted and > the other side is public. This problem is cause because the SDP info only is > present in the 200 OK and in the ACK message. > > I'm using ser (8.14) with mediaproxy (1.2.1). > > I have an H323 to SIP translator (and SIP to H323 too) connected to the > PSTN and to the SIP Proxy (SER). > H323 is the origin of the call (public client), and SIP UAC is the > destination (Natted client). > My proxy is working in state full mode and has a public IP. > > The H323 side is using slow start, so when it starts the transaction with > an INVITE, it will not send the SDP info (it will send it in the ACK > message). > The proxy forwards this message to the destination. > SIP UAC, starts ringing. When we hooks off, the SIP UAC send an 200 OK > with SDP message to the H323 side. > > This is the 200 OK / SDP that the proxy receives from the SIP side. > > No. Time Source Destination Protocol > Info > 488 28.488811 200.68.89.2 200.68.89.12 SIP/SDP > Status: 200 OK, with session description > Message body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): 700600 6015 6015 IN IP4 > 192.168.0.101 > Owner Username: 700600 > Session ID: 6015 > Session Version: 6015 > Owner Network Type: IN > Owner Address Type: IP4 > Owner Address: 192.168.0.101 > Session Name (s): AddPac Gateway SDP > Connection Information (c): IN IP4 192.168.0.101 > Connection Network Type: IN > Connection Address Type: IP4 > Connection Address: 192.168.0.101 > Time Description, active time (t): 0 0 > Session Start Time: 0 > Session Stop Time: 0 > Media Description, name and address (m): audio 23018 RTP/AVP > 18 8 0 101 > Media Type: audio > Media Port: 23018 > Media Proto: RTP/AVP > Media Format: ITU-T G.729 > Media Format: ITU-T G.711 PCMA > Media Format: ITU-T G.711 PCMU > Media Format: 101 > > Following the ser.cfg, this 200 OK / SDP will make the proxy to use > mediaproxy (use_media_proxy()), because the destination user is natted. So > the mediaproxy module, will generate a lookup command to the > proxydispatcher.py. > > proxydispatcher[30535]: command lookup 2281401749(a)200.68.89.10 > 192.168.0.101:23018:audio 200.68.89.2 200.68.89.10 remote 200.68.89.10unknown AddPac=20SIP=20Gateway info= > from:1150316660@200.68.89.10,to:1152464490@200.68.89.10 > ,fromtag:3512844671,totag:7b07717a4 > proxydispatcher[30535]: warning: trying to lookup session with > non-existent id: '2281401749(a)200.68.89.10' > > The proxydispatcher does not recognize that command because it has not > generate the session before. So, the proxy forwards the 200 OK /SDP message > without changing the SDP info. When the H323 sides receives that info, it > thinks that it has to send RTP to the private IP. > > Then the H323 sends the ACK with SDP. > > Then, we cannot ear audio in both sides. > > Here is the problem, because, I didn't receive an INVITE with SDP that can > create the session into the dispatcher. > > > A is public, B is natted > > side A Proxy side B > > INVITE without SDP > ----------> > 100 Trying > <---------- > > INVITE without SDP > ----------> > > 100 Trying > <---------- > 180 Ringing > <---------- > 180 Ringing > <---------- > > 200 OK SDP (B private IP) > <---------- > *1* > 200 OK SDP (B private IP) > <---------- > > ACK SDP (A public IP) > ----------> > *2* > ACK SDP (mediaproxy public IP) > ----------> > > ...................................................... > RTP from B > <---------------------- > *3* > ...................................................... > > > *1* The proxy must replace the private IP of the SDP, with the ip of the > mediaproxy. The looku prequest does not work, because no session was found > *2* the proxy generates the session in the mediaproxy. So B thinks that > the RTP must go to the mediaproxy. But A never knows the mediaproxy address. > *3* RTP from A goes to anywhere, because A does not know where it is the > private adrress of B. > > Do you know a solution to this? Is the sequence of the message all right? Regards. Carla

18 years, 9 months

[Serusers] Here I go again...Call type processing Section problem?

by Juan

Hi. As I had no answers to this question, here it is again. Till now I was happy using SER handling communications between UAs using nathelper and RPT proxy. Now I changed the ser.cfg script to the one with PSTN support and none of the UAs get registered. I am getting a 403 Forbidden. Analyzing the script I understood that this 403 was coming from the Call Type Processing Section so I changed if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { and so on to if (uri!=myself) and now the UAs are registering again. But of course I can't make calls. So I would like to know how this< is_uri_host_local and is_from _local> work and what should I do to make them allow my UAs to register. I am attaching the ser.cfg that doesn't work. Thanks in advance Juan Ferrari

18 years, 9 months

[Users] TM bug?

by Greg Fausak

I am running a proxy pushing 2 million calls a day. Every now and then, I drop 5 or 10 BYE messages. I've attached the sip_scenario log file, F16,F17,F20 all send BYE messages to 12.46.104.62. F21 is sending the BYE form 12.46.104.62 to 12.46.104.252. The OK comes back, but is never relayed. BYEs continue to retry until 12.46.104.62 finally gives up and sends a Request timeout 408 in frame F38. So, F38 is evidence that the transaction started. Right? Why are all of the BYEs ignored, and why isn't the OK relayed? I have xlog output from 12.46.104.58 indicating each time a BYE is loose-routed to 12.46.104.62. The syslog file shows each attempt (F15,F18,F19,F23, etc). I also have xlog output from 12.46.104.62, however, the syslog on that machine never shows receiving or loose-routing a BYE with this callid. It seems like receiving more than one BYE in a short period of time is screwing up the t_relay()??? The calls are being generated with sipp, and 999,990 times out of a million it works. ---greg -- Greg Fausak greg(a)thursday.com

18 years, 9 months

Re: [Fwd: [Users] enum patch]

by Greg Fausak

Juha, I've updated the documentation. I don't know how to produce output, so I don't know if it works. I haven't done a patch before either, so, let me know if this is the preferred format or if you want to see it a different way. By the way, when I was reading the documentation I noticed that the enum query puts the highest priority query in the ruri, then the rest of them are stored in new branches with a q value. I haven't actually tried this. Is it the plan that these branches will be looped through serially upon failure without having a retry route block? The new patch is attached. Thanks, -g On Jan 29, 2006, at 4:27 AM, Bogdan-Andrei Iancu wrote: > To be more precisely, the file is modules/enum/doc/enum_user.sgml > > bogdan > > Juha Heinanen wrote: > >>> is the documentation the README file? >>> >> >> you have to edit the xml file from which README is generated >> automatically. >> >> -- juha >> >> >

18 years, 9 months

[Users] dispatcher module

by Jayesh Nambiar

Hi all, I am trying to use the dispatcher module. I have two gateway address in the dispatcher.list file. But when it reaches that particular gateway, it does not rewrite the host with that gateway. Is dispatcher not supposed to do it?? One more thing.... when I used the function forward(); for forwarding the call to set of gateways, the openser shows an error saying that it is "bad forwardarguement". openser needs the ip addr and the port in the forward function, but then how will it look at the destination set. It worked when I used t_relay(), but I need to forward it statelessly. I just used the function as it was shown in the example of the dispatcher module in the docs.. can someone please help me on this.. Thanks in advance. jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket

18 years, 9 months

[Users] Using uac_replace_from to munge username between ser and Asterisk?

by Barry Flanagan

Hi, I am trying to get a multi-domain setup working between OpenSER and Asterisk (PSTN gateway and Voicemail). It appears that I need to munge the username before it goes to Asterisk so that it includes the domain element, otherwise there is clashing in the username namespace on Asterisk. It looks like I should be able to use uac_replace_from to change the From URI from user(a)domain.com to user_domain(a)domain.com so that then I can set up the users in Asterisk as [user_domain] Does this sound sane, and could someone tell me how I can accomplish this in OpenSER? I have tried a few things but they all end up as bad headers. Thanks in advance. -- -Barry Flanagan

18 years, 9 months

[Serusers] Presence + XCAP root problem with multiple domains

by Adrian Georgescu

Hello Vaclav, I cannot manage to handle the same username under different domain names using the same XCAP tree. Data location for user1(a)example1.com will conflict with data location for user1(a)example2.com on the filesystem is the same: /xcap-root/pres-rules/users/user1/presence-rules.xml My XCAP server can generate separate trees for each domain and create the XCAP structure under each domain (like /xcap-root/example1/pres-rules/users/ ) but in this case SER should be able to build a dynamic root like: https://xcap.example.com/$domain$/ In Eyebeam I can configure the xcap-root statically but obviously if I do this for multiple domains, SER should be able to build it from information available in the PUBLISH/SUBSCRIBE request so that it can locate the right tree on the XCAP server. Do you see any other solution for this problem? Regards, Adrian

18 years, 9 months

[Serusers] Serweb does not send mail on registration

by Bart J. Smit

I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and ser-0.9.6-1 compiled from the source RPM at http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new user, the PHP script goes to an empty page without title and the email never gets sent to the address entered on the registration form. The database gets the correct user details inserted into the pending table. The sendnotification and greeting fields remain empty. I have modified the line in config_data_layer.php for the collation character set to read: $config->data_sql->collation = "latin1_swedish_ci"; This got rid of a DB error in serweb which I traced through pear logging: Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and (utf8_general_ci,COERCIBLE) for operation '='] I have sent a test email with the debugging/mail.php script and it arrives fine. I have tested the registration page with local recipients, recipients on the LAN and external recipients but the behaviour stays the same. All the pages I have googled on serweb registration problems always state that the emails are sent out correctly, so I'm likely missing something obvious. Can anybody help me get the registration to work? Thanks, Bart...

18 years, 9 months

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sr-users January 2006