search for: sipuri

Hi all, I have a need to strip some characters from a variable to get the right data but have only found how to strip all but the last or middle stuff, need to keep the beginning. EG: With $(SIPURI) I want to keep just the sip number and delete the remainder '@server.com'. Ideally I'd like to use 'SayDigits($(sipuri[-@server.com])' All replies greatfully accepted. Phil --------------------------------------------------------------------- To unsubscribe, e-mail: asteri...

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

It there a native asterisk dialplan function which will tell me the position of a specific character in a given string? eg if I wanted to find what position the '@' was at in ${SIPURI} Thanks in advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH...

Hi We're using asterisk 1.4.17 debian package soon moving to 1.8 rpm package. When using MixMonitor to do call recordings, for outbound calls I have been using the channel variable SIPURI to get the originating SIP extension name. I have now stumbled across a few files where the SIP extension name must be incorrect when cross referencing the call with other sources (such as the channel shown in the cdr). So, a couple of questions I'm throwing out there: Why would the Channel va...

...Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=XXXXXXXXX SIPURI=sip:303 at XXXXXXXXXXXXXXXXX CDR Variables: level 1: clid="Ext. 303" <303> level 1: src=303 level 1: dst=XXXXXXXXXX level 1: dcontext=XXXXXXXXXXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 l...

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

...f ---------------------------- [default] exten => 1000,1,Dial(SIP/alex||t) [sip-in] exten => 1000,1,Dial(SIP/alex||t) [outgoing] exten => _0.,1,Dial(SIP/rix/${EXTEN}|20|t) ---------------------------- .qt/kphonerc ---------------------------- [Registration] AutoRegister=No SipServer= SipUri="Alex Polite" <sip:alex@localhost> UserName=alex qValue= ---------------------------- -- Alex Polite http://polite.se

...oicepulse-in Extension= 12222222 Priority= 4 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: SIPCALLID=282e93ca78805a039fdf01729af52c at 64.62.94.171 SIPUSERAGENT=Asterisk PBX SIPDOMAIN=66.195.225.160 SIPURI=sip:3333333 at 64.62.94.171 <sip%3A3333333 at 64.62.94.171> ================================================================================ -- Executing [12222222 at voicepulse-in:5] VoiceMail("SIP/mrXXXX-08XXXX", "101 at default|u") in new stack -- <SIP/mrXXXX...

> Can anyone help me to solve the next problem???? > I need to get the caller's identity from the Sip messages. update asterisk with the CVS head and use ${SIPURI } funziona bene Sergio

...39;t seem to be the case, however. Here is a bit of the verbose console output: (Please note that I added some extra ast_log calls to the source code to generate some extra debugging information.) Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPURI, value=sip:nyct-901@192.168.74.33:5060 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPUSERAGENT, value=Polycom...

...the 'ext' number, not the incoming SIP callerid as can be seen on incoming calls when I register the phone directly to provider. I tried to add 'o' option to Dial but same results. The closest thing (and perhaps not the smartest) I could do: exten => ext,1,Set(CALLERID(name)=${SIPURI}) exten => ext,2,Dial(SIP/phone1&SIP/phone2) If that matters, phone1 and phone2 are Grandstream HTs with callerid-capable cordless phones connected to FXS ports. How can I get the calling number (and perhaps a hint to distinguish the SIP account for incoming call) to be displayed on those...

Hi List, Which will be the appropriate variable to get called peer IP address? I tried following channel variables peerip, recvip, URI, from and following SIP channel variables: SIPURI,SIPDOMAIN They all return calling peer IP but not the destination/called peer IP. unfortunately set(CDR(calledip)=${CHANNEL(to)}) doesn't work Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-...

...AME=20230505110016-customer-DE-EXTEN-49xxxxxxx-CLINUM-49xxxxxxxx-CLINAME--PAICLEAN--CLICLEAN-49xxxxxxxxxx-OCLINUM--OCLINAME- SIPADDHEADER01=P-Asserted-Identity: <sip:+49xxxxxxxx at x.x.x.x> CLICLEAN=49xxxxxxxxxx CLILEN=12 SIPCALLID=85b9164eeb2211eda29c008cfa0447f8 at x.x.x.x SIPDOMAIN=x.x.x.x SIPURI=sip:49xxxxxxxxx at x.x.x.x:5061 CDR Variables: level 1: customer=customer level 1: country=DE level 1: dnid=49xxxxxxxx level 1: clid="" <49xxxxxxxxx> level 1: src=49xxxxxxxx level 1: dst=49xxxxxxxx level 1: dcontext=customer-voipin level 1: channel=SIP/customer01-0000dfa4 level 1...

...appreciated, Derek zaptel.conf: span=1,1,0,esf,b8zs e&m=1-24 loadzone = us defaultzone=us zapata.conf: [channels] language=en context=from-pstn signalling = em_w rxgain=2 group = 0 channel => 1-24 /var/log/asterisk/full (snip): Jun 7 08:15:09 DEBUG[14754] channel.c: Not copying variable SIPURI. Jun 7 08:15:09 DEBUG[14758] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jun 7 08:15:09 DEBUG[14754] chan_zap.c: Dialing '(snipped)' Jun 7 08:15:09 DEBUG[14754] chan_zap.c: Deferring dial...

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

...rom_customers-54-1. Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPCALLID. Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPUSERAGENT. Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPDOMAIN. Feb 13 23:50:35 DEBUG[28047] channel.c: Not copying variable SIPURI. Feb 13 23:50:35 DEBUG[28049] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Feb 13 23:50:35 VERBOSE[28047] logger.c: -- Requested transfer capability: 0x00 - SPEECH Feb 13 23:50:35 VERBOSE[28047] logger.c: -- Called g1/54 Feb 13 23:50:35 DEBUG[28047] channel.c:...

...G[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Mar 26 14:36:38] DEBUG[12577]: channel.c:3766 ast_channel_inherit_variables: Not copying variable SIPURI. [Mar 26 14:36:38] DEBUG[12577]: chan_dahdi.c:2301 dahdi_call: Dialing '8685XXXXX' [Mar 26 14:36:38] DEBUG[12577]: chan_dahdi.c:2379 dahdi_call: Deferring dialing... (res -1) [Mar 26 14:36:38] DEBUG[11975]: channel.c:1133 channel_find_locked: Avoiding initial deadlock for channel '0x24a...

...el.c: Not copying variable DIALEDPEERNAME. 13:37:40 channel.c: Not copying variable DIALEDPEERNUMBER. 13:37:40 channel.c: Not copying variable DIALSTATUS. 13:37:40 channel.c: Not copying variable SIPCALLID. 13:37:40 channel.c: Not copying variable SIPDOMAIN. 13:37:40 channel.c: Not copying variable SIPURI. 13:37:40 chan_mobile.c: Calling Carlos/909037079681 on Mobile/Carlos-0213 13:37:40 ] app_dial.c: -- Called Carlos/909037079681 13:37:40 channel.c: Set channel Mobile/Carlos-0213 to read format ulaw 13:37:40 channel.c: Set channel SIP/1000-0021a568 to read format slin 13:37:40 chan_mobile.c: **...

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

...nal-003381765432-1. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable SIPCALLID. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable SIPUSERAGENT. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable SIPDOMAIN. Mar 31 16:53:57 DEBUG[11747] channel.c: Not copying variable SIPURI. Mar 31 16:53:57 DEBUG[24349] devicestate.c: Changing state for Zap/1 - state 2 (In use) Mar 31 16:53:57 VERBOSE[11747] logger.c: -- Requested transfer capability: 0x00 - SPEECH Mar 31 16:53:57 DEBUG[11751] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Mar 31 16:5...