sr-users April 2006

sr-users@lists.kamailio.org

188 participants
431 discussions

[Users] How to extract sip-id when using alias-id?

by Frogger

I am using alias_db to store alias's for users. However when an alias is dialed I need to also be able to restore the alias owners original sip-id. This is necessary to forward the call to voicemail using sip-id. Any guidance on getting the sip-id out of the alias_db and replacing the alias-id prior to forwarding to vmail? Thanks in advance! F __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 5 months

[Users] 401 Unauthorized

by Joao Pereira

Hello to all Im trying to register SIP clients (Cisco IP phone and Planet VIP )that are located behind an ADSL line ( and behind NAT), and the first time they register, they receive the OK from OpenSER, but when I reboot the phone, they cant connect again. They receive "401 Unauthorized" from OpenSER. OpenSER isnt returning "401 Unauthorized" in any part of openser.cfg Im not using firewall. Can the problem be in the type of NAT? Can someone help me? Thanks Joao Pereira

18 years, 6 months

Re: [Serusers] I have problem with my SERWEB version 0.9.4 (Installed on FC4) , with SER version is 0.9.6

by Karel Kozlik

pear-log package is not probably installed. If it is, see php log or apache error log. Karel Phuong Nguyen Trong napsal(a): > When I enable logging in serweb, and then, I try to access to SERweb home > page( http://mydomain.com/serweb/admin/index.php ). > There is a white page (It means that on screen don't have anything!). > But, when I disable logging, the "log in" web page is display OK. > Could you find any problems on this? > > Phuong Nguyen > IP Clients Group > ESCS Dept > TMA company. > http://www.tma.com.vn/ > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: Wednesday, April 19, 2006 12:31 PM > To: Phuong Nguyen Trong > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] I have problem with my SERWEB version 0.9.4 > (Installed on FC4) , with SER version is 0.9.6 > > Hi, > enable logging in serweb, set log level to debug and see the log file. > > Karel > > Phuong Nguyen Trong napsal(a): >> I try to install SERweb 0.9.4 is running with SER 0.9.6 ( Is it OK?) >> SER is running OK but I don't have software to test the connection is >> managed by SER. Anyone have software to test its? >> >> My major problem is: SERWeb is running OK with user. I can create, update, >> delete user using account of normal user. >> But I cannot access to admin mode using default username ( admin ) and >> default password ( heslo ). The phenomenon of error is: (Only one line to >> describe this error). >> Bad username or password >> Only one line to describe this error. >> I try to solve this problem rely on previous resolved similar problems in >> our mailing lists. But it still have this problem. >> Thanks a lot for any solution of this, >> >> Phuong Nguyen >> >> IP Clients Group >> ESCS Dept >> TMA company. >> http://www.tma.com.vn/ >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >

18 years, 6 months

[Users] call forwarding and billing

by Jayesh Nambiar

Hi folks, I have implemented the openser.cfg with forwarding and voicemail features completely. Only problem is, if the user is forwarding the call on a pstn number how should he be billed, as the one who has forwarded is accountable and not the one who called. I have implemented forwarding using avps serial forking method. I've read about multiple leg accounting and was not successful in understanding it properly. The docs only mention to set the parametrs and after I set the params, I get a n/a in src_leg and dst_leg column. Can someone pls give me examples of successfully using multiple leg accounting in openser.cfg. Is it like, the call leg accounting will only take place if there is a 3XX response from the server. I am asking this as my cfg does not produce any 302 response, but creates a brand new Invite request for the forwarded call. Can someone pls tell me more about multi-leg accounting. Also I read about the diversion module which can be used in the forwarding scenarios. But I added the diversion header and forwarded the call to Cisco gateway, it gave a 400 Bad Request reponse saying 'Malformed CC-Diversion/Diversion/CC-Redirect Header'. Can someone explain what does this mean. It will help me a lot as this is the only thing I am left with. Thanks a lot in advance. w/regards, Jayesh --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

18 years, 6 months

[Users] OpenSER Docs

by Douglas Garstang

Has anyone seen any good OpenSER documentation, or know of any training courses in North America that relate directly to OpenSER? I've been involved with OpenSER for about 6 months now and the available documentation has remained about the same - REALLY BAD. Thanks, Doug

18 years, 6 months

[Users] clustering

by unplug

How to do the clustering of openser? Any reference?

18 years, 6 months

[Users] NEW MODULE : dialog support

by Bogdan-Andrei Iancu

Hi, the devel version includes a new module for providing dialog support. For more about it, please see: http://openser.org/docs/modules/1.1.x/dialog.html Currently the module offers only call tracking functionality. In the future, new functionalities as dialog termination, per dialog flags and AVPs will be added. These will make possible implementation of more complex dialog-based features (like location hiding). regards, bogdan

18 years, 6 months

[Serusers] Modifying "From" header field of an INVITE message

by scm-j＠nuntius.com

I am relatively new to SER. My SIP service provider requires me to present the ANI in the "From" field in my INVITE message header in E.164 format. I am currently sending this information as follows: From: "2125551212" <sip:2125551212@xxx.xxx.xxx.xxx> They want it in E.164 format as follows: From: "+12125551212" <sip:+12125551212@xxx.xxx.xxx.xxx> I see function only to modify the SIP URI but not the other header or SDP fields. Could someone help me with a code snippet that I can use in ser.cfg to do this conversion? Regards, SCM

18 years, 6 months

[Users] usrloc and cseq

by Helge Waastad

Hi, I have a Loadbalanced scenario with two servers of each (2x LB/Dispatchers and 2x Routers) Each router has its own location table. They both replicate REGISTERS (forward_tcp/save_noreply). However every now and then I get: ERROR:usrloc:update_contacts: invalid cseq for aor This happens for all contacts I do not really know why, since all the transactions seems allright to me and that it happens only now and then... Any clues to help med debug further? br hw -- Helge Waastad Senior Konsulent Smartnet

18 years, 6 months

[Users] Attended Transfer

by Bastian Schern

Hello, does anybody got a working configuration to make an "attended call transfer" with a call through an Asterisk gateway? Example: PSTN --> Asterisk --> SER --+-- A | +-- B The call will come from the PSTN Network and will go through "A". A sets the call on "Hold" and calls "B". After A is connected with B, A hangup an B got the call from PSTN. This in _not_ working at the moment. Attended call transfer only with OpenSER and only SIP-Phones is no Problem. But if the is an Asterisk as PSTN-GW in the game it will not work. Regards Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.7010 from 25.04.2006 Virus news: www.antiviruslab.com

18 years, 6 months

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sr-users April 2006