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 26 May '06

26 May '06

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

2 3

[Users] 401 Unauthorized
by Joao Pereira 18 May '06

18 May '06

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

7 11

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 15 May '06

15 May '06

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 &… [View More]

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[Users] call forwarding and billing
by Jayesh Nambiar 10 May '06

10 May '06

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 … [View More]

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[Users] OpenSER Docs
by Douglas Garstang 10 May '06

10 May '06

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

3 3

[Users] clustering
by unplug 10 May '06

10 May '06

How to do the clustering of openser? Any reference?

2 1

[Users] NEW MODULE : dialog support
by Bogdan-Andrei Iancu 09 May '06

09 May '06

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

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[Serusers] Modifying "From" header field of an INVITE message
by scm-j＠nuntius.com 04 May '06

04 May '06

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 … [View More]

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[Users] usrloc and cseq
by Helge Waastad 03 May '06

03 May '06

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

2 1

[Users] Attended Transfer
by Bastian Schern 02 May '06

02 May '06

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 … [View More]

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sr-users April 2006