search for: cid_num

...all.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to do this. { struct ast_channel *callbk; char *callbk_real_context; char xferto[256],dialstr[265]; char *cid_num; char *cid_name; int outstate=0; char *exten = NULL ,*context = NULL; pu = head; //pu is a queue hav dst and src number printfl("\n\n\n\n %s time is over",pu->dst);...

Hello, How can I access caller's number with Asterisk CVS 1.0.12? In new version there are structure cid with field cid_num. And in 1.0.12 only callerid field which is equal to cid_name. I also tried to get it from chan->cdr->src but this is also the same as cid_name or callerid. Mindaugas Kezys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.dig...

Hi All, I am trying to add FAX to my SIP confiig and I am getting some errors, any help would be great. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-v1-0-12/23/04-22:36:11\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

...snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars = ast_variable_new("_PARKEDAT", buf); dchan = __ast_request_and_dial(dialtech, AST_FORMAT_SLINEAR, dialstr,30000, &outstate, chan->cid.cid_num, chan->cid.cid_name, &oh); I assume (I hope not incorrectly) that I have to modify the variable chan->cid.cid_name Could one of the Asterisk gurus point me in the right direction as to how to do this? Thanks in advance Brian -------------- next part -------------- An HTML attachment w...

...quot;, "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name, dst->name, src->cid.cid_num ? src->cid.cid_num : "<unknown>", src->cid.cid_name ? src->cid.cid_name : "<unknown>", src->uniqueid, dst->uniqueid, (dst->cdr)?(dst->cdr->userfield):""); } I am writing this mail from home, so don't really have t...

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

...le phones ; 08/2005 Stefan Gofferje [lines] id = 6000 pin = 1234 label = 6000 description = Office context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Office cid_num = 6000 line => 6000 id = 6001 pin = 1234 label = 6001 description = LivingRoom context = client_int_unrestricted callwaiting = 1 incominglimit = 2 mailbox = 1000 vmnum = 8500 cid_name = Living Room cis_nu...

...= 102,User2,102@wct-internal speeddial = 103,User3,103@wct-internal speeddial = 104,User4,104@wct-internal device => SEP000F90CEF9D3 [lines] id = 101 pin = 1234 label = User1 description = Line101 context = wct-internal incominglimit = 2 mailbox = 101@wct-internal vmnum = 8501 cid_name = User1 cid_num = 101 line => 101 id = 401 pin = 1234 label = Tech Support description = Line401 context = webcore-internal incominglimit = 2 mailbox = 400@webcore-internal vmnum = cid_name = Tech Support cid_num = 401 line => 401 --------------------------------------------- extensions....

...eaving directory `/usr/src/asterisk/apps' > make: *** [subdirs] Error 1 > > Regards, > Kerry For a while, the structure switched over. If you are getting an error with "callerid" in your build, try searching the code (app_rxfax.c) and replace the occurances with "cid.cid_num". Justin Newman Newman Telecom, Inc.

.... Flow enters in zt_call and generates callerid id of length 8867 from callerid generate in callerid.c *****************snip** zt_call** chan_zap.c************************** if (p->cidspill) p->cidlen = ast_callerid_generate(p->cidspill, ast->cid.cid_name, ast->cid.cid_num, AST_LAW(p)); p->cidpos = 0; send_callerid(p); ************************************************************************ //Flow enters in send callerid in a while loop which checks cidpos<cidlen; Initial cidpos=0 and cidlen =8867 ***************snip** s...

...; future use label = 2002 ; button line label description = Line 2002 ; top diplay description context = from-sccp-intenal incominglimit = 2 transfer = on mailbox = 1001 vmnum = 2999 cid_name = Phone2002 ; caller id name cid_num = 2002 trnsfvm = 1000 secondary_dialtone_digits = 9 secondary_dialtone_tone = 0x22 ; outside dialtone musicclass=default language=en rtptos = 18 echocancel = on silencesuppression = off line => 2002 extensions.conf [from-sccp-internal] include => local-extensions include => always-out-...

...o2 tzoffset = 0 trustphomeip = no autologin = 105 device => SEP00036B0123 [lines] id = 104 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 104 vmnum = 500 cid_name = Cisco1 cid_num = 104 line => 104

...E[952]: res_odbc.c:1502 odbc_obj_connect: res_odbc: Connected to PostgreSQL-asterisk [asterisk-connector] [Oct 22 21:46:09] WARNING[952]: cel_odbc.c:733 odbc_log: Insert failed on 'PostgreSQL-asterisk:celnew'. CEL failed: INSERT INTO celnew (uniqueid,linkedid,eventtime,userdeftype,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,accountcode,peeraccount,userfield,peer,amaflags) VALUES ('co-1287780365.0','co-1287780365.0',{ ts '2010-10-22 21:46:06' },'','135','441XXXXXXXXX','441XXXXXXXXX','','...

...t,Custom:q8501_a8512 [macro-custom-agent-inout] ; ; Standard extension macro: ; ${ARG1} - Queue to Join ; exten => s,1,Answer() exten => s,n,MYSQL(Connect connid localhost asterisk xxxxxx yyyyy) exten => s,n,MYSQL(Query resultid ${connid} SELECT channel, extension, name FROM pbx WHERE cid_num='${MACRO_EXTEN:4}') exten => s,n,MYSQL(Fetch fetchid ${resultid} channelpath CALLBACKNUM callername) exten => s,n,MYSQL(Clear ${resultid}) exten => s,n,MYSQL(Disconnect ${connid}) exten => s,n,AddQueueMember(${ARG1},SIP/${MACRO_EXTEN:4}) ;If they're already logged in, log o...

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

...= 0 autologin = 104 ; speeddial = 101, 105 device => SEP00036BC3852B [lines] id = Cisco1 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 500 vmnum = 500 cid_name = Cisco1 cid_num = 104 line => 104 extensions.conf [internal] include => outbound-local include => outbound-long-distance ; Software phone exten => 101,1,Dial(SIP/test-softphone,,r) exten => 102,1,Dial(SIP/bob,20) exten => 102,2,Voicemail(u102) exten => 102,102,Voicemail(b102)...

...OOK, &x); if (res) { fprintf(stderr, "Unable to go offhook...\n"); } fprintf(stderr, "Polarity should be reversed now...\n"); dop.op = ZT_DIAL_OP_REPLACE; // if (ast->cid.cid_num) { snprintf(dop.dialstr, sizeof(dop.dialstr), "TwD1234567890CC"); // } else { // snprintf(p->dop.dialstr, sizeof(p->dop.dialstr), "TB00Cw"); // } if (ioctl(fd, ZT_DIAL, &dop)) {...

...; per line transfer capability. on, off, 1, 0 mailbox = 30 ; voicemail.conf (syntax: vmbox[@context][:folder]) vmnum = *97 ; speeddial for voicemail administration, just a number to dial cid_name = JJJ ; caller id name cid_num = 30 trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail) secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits) secondary_dialtone_tone = 0x21 ; outside dialtone musicclass=default ; Sets the defaul...