Guys any advice please... Thx.
> Just prompt explaination: >> - no modparams in config > no uac modparams in config. > > > >> - route[0] is responsible for basic routing >> - route[5] is for proper call distribution by using lookup(location) >> information. In the same route i have implemented calling number >> modification. >> Just before the end of route i am arming t_relay with failure route >> (in case of busy for instance). >> - failure_route[105] is to do_routing the call to VM service outside >> the opensips. But just before t_relaying here i need to restore the >> original $fU. >> According to >> http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id292928 >> i should use uac_restore_from() command (with default restore_mode >> modparam). >> Unfortunately the calling number once replaced cannot be restored in my case. >> Below you may find a snippet from debug >> >> Number replacing is generating vsf param >> /sbin/opensips[28268]: DBG:uac:w_replace_from: dsp=0xffd3f38c (len=0) >> , uri=0xffd3f394 (len=29) >> /sbin/opensips[28268]: DBG:uac:replace_uri: removing display ["unknown"] >> /sbin/opensips[28268]: DBG:uac:replace_uri: uri to replace >> [sip:[email protected]] >> /sbin/opensips[28268]: DBG:uac:replace_uri: replacement uri is >> [sip:[email protected]] >> /sbin/opensips[28268]: DBG:uac:replace_uri: encode >> is=<AAAAAAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--> len=44 >> /sbin/opensips[28268]: DBG:rr:add_rr_param: adding >> (;vsf=AAAAAAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--) 0x81c2828 >> >> then opensips constucts Busy message and in the same time without any >> uac_restore_from() command: >> /sbin/opensips[28795]: DBG:uac:restore_uri_reply: removing >> <<sip:[email protected]>> >> /sbin/opensips[28795]: DBG:uac:restore_uri_reply: inserting <"unknown" >> <sip:[email protected]>> >> >> then the call failes to failure_route[105] and uac_restore_from() is >> generating following debug >> /sbin/opensips[28797]: DBG:uac:restore_uri: getting 'vsf' Route param >> /sbin/opensips[28797]: DBG:uac:restore_uri: route param 'vsf' not found >> just after that the call is hitting do_routing and t_relay to VM >> server. Of course calling number was not restored to original one. >> >> Could You please point me where the problem is located? >> Just one more info - calling number modification part of config is >> located in separated route[10] to be used whenever i wish in my >> script. >> >> Thx, >> Maciej. >> >> >>> Bogdan, Stefano, >>> >>> Its working as is should :) >>> Thanks for pointing me to the right function. >>> >>> Maciej. >>> >>> 2010/10/11 Bogdan-Andrei Iancu <[email protected]>: >>>> To be more precise: >>>> http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582 >>>> >>>> Regards, >>>> Bogdan >>>> >>>> Stefano Pisani wrote: >>>>> Use replace_from :-) >>>>> >>>>> ciao >>>>> s >>>>> >>>>> Il 10/10/2010 19:19, Maciej Bylica ha scritto: >>>>> >>>>>> Hello >>>>>> >>>>>> I have a question regarding $fU pseudo variable. >>>>>> As mentioned at http://www.opensips.org/Resources/DocsCoreVar16 and on >>>>>> the basis of opensips outputs: >>>>>> ERROR:dialplan:dp_trans_fixup: the output PV is read-only!! >>>>>> it clearly means that $fU is read-only. >>>>>> >>>>>> Unfortunately it is quite big problem for me, because what im >>>>>> struggling with is to achieve proper calling number presentation. >>>>>> In my scenario all endpoints located in subscriber table do have full >>>>>> username with country code, so there are for instance: >>>>>> - 48111223344 (48 country code) >>>>>> - 49222334455 (49 country code) >>>>>> - 44333445566 (44 country code) >>>>>> ... >>>>>> >>>>>> If there is a national call inside the 48 country code the calling >>>>>> number should be changed by striping first two digits (48) - >>>>>> 48999887766--->999887766 >>>>>> In case of international call, i should add two digits (00) - >>>>>> 49222334455--->0049222334455. >>>>>> >>>>>> I am using diaplan module in this case and following entry gives me >>>>>> the error I mentioned. >>>>>> dp_translate("2", "$fU/$fU"); >>>>>> >>>>>> If there are any workaround. >>>>>> Any help would be highly appreaciated. >>>>>> >>>>>> Thanks, >>>>>> Maciej >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> [email protected] >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> [email protected] >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> >>>> -- >>>> Bogdan-Andrei Iancu >>>> OpenSIPS Bootcamp >>>> 15 - 19 November 2010, Edison, New Jersey, USA >>>> www.voice-system.ro >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
