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...323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com] Sent: 01 December 2003 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How do I get caller's number in oh323 ? We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via...

...list, I purchased a couple for evaluation. They do work with asterisk - and are good value for money, but as somebody commented: they are not yet perfect. I just wondered if anybody had managed to get either the message-waiting indicator or the conference button to work? Phil Skuse <phil.skuse@vicorp.com> **************************************************** Unix System Administrator, Vicorp Group Limited. Tel +44 (0)1753 660523 Fax +44 (0)1753 660501 The Telephony Engine Company http://www.vicorp.com ****************************************************

...0x00022012 h323.cxx(4594) H323 SendUserInputIndicationQ931("1") My initial examination of the code suggests that it should be calling SendUserInputAsTone instead. (Please don't use this thread as an excuse to start the h323 vs oh323 war AGAIN) Phil Skuse <phil.skuse@vicorp.com> *************************************************** UNIX System Administrator. NIC Handle: MBJEJPIEUI Vicorp UK Limited: The Telephony Engine Company. Tel +44 (0)1753 660523 http://www.vicorp.com ***************************************************

...ort EXTENSION=$VARVALUE;; ("agi_callerid") export CALLERID=$VARVALUE esac read STDIN done echo `date` Extension: $EXTENSION CallerID: $CALLERID >> /home/asterisk/call_log.txt echo "CallerID: $CALLERID" | mail -s "Somebody called the demo" -c phil@vicorp.com Here is extensions.conf... exten => 606,1,AGI,call_logger.sh exten => 606,2,Dial,sip/606|10|t exten => 606,3,HangUp Phil Skuse <phil.skuse@vicorp.com> *************************************************** UNIX System Administrator. NIC Handle: MBJEJPIEUI Vicorp...

...dge to interpret the log file, could somebody please give me a clue? [2004/10/27 17:36:41, 5] smbd/posix_acls.c:(903) unpack_nt_owners: validating owner_sids. [2004/10/27 17:36:41, 10] nsswitch/wb_client.c:(98) winbind_lookup_sid: SUCCESS: SID S-1-5-21-555165157-1185895069-398547282-1628 -> VICORP Nick.Button [2004/10/27 17:36:41, 10] passdb/lookup_sid.c:(418) sid_to_uid: winbind failed to allocate a new uid for sid S-1-5-21-555165157-11 85895069-398547282-1628 [2004/10/27 17:36:41, 3] smbd/posix_acls.c:(921) unpack_nt_owners: unable to validate owner sid for S-1-5-21-555165157-11858950...

...;m guessing that I either need to collect the DTMF, format it into a sip address and then somehow get asterisk to dial that address, or perhaps I can take it from $EXTEN somehow. Is this possible with the existing apps/scripts/macros, or do I need to write some new ones? Phil Skuse <phil.skuse@vicorp.com> **************************************************** UNIX System Administrator, Vicorp Group Limited. Tel +44 (0)1753 660523 Fax +44 (0)1753 660501 The Telephony Engine Company http://www.vicorp.com ****************************************************

...n't get the number. How can I make this work? h323.conf ======= [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf =========== exten => 602,1,Dial,h323/7002@ivr exten => 602,2,HangUp Phil Skuse <phil.skuse@vicorp.com> *************************************************** UNIX System Administrator. NIC Handle: MBJEJPIEUI Vicorp UK Limited: The Telephony Engine Company. Tel +44 (0)1753 660523 http://www.vicorp.com ***************************************************

...ugh Xlite? The SIP messages all look OK to me, but the sound isn't coming through. It was trying to use GSM, so I searched the archive and tried: disallow=gsm allow=ulaw Now it says that it's using ULAW but I still get no sound in either direction. Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com> **************************************************** UNIX System Administrator, Vicorp Group Limited. Tel +44 (0)1753 660523 Fax +44 (0)1753 660501 The Telephony Engine Company http://www.vicorp.com ****************************************************

...ybody have an example of this? Are there any particular security risks that I need to defend against? Would it be better to put a secured asterisk server outside the firewall and connect it to the internal one with IAX? Does this require less ports open on the firewall? Phil Skuse <phil.skuse@vicorp.com> **************************************************** Unix System Administrator, Vicorp Group Limited. Tel +44 (0)1753 660523 Fax +44 (0)1753 660501 The Telephony Engine Company http://www.vicorp.com ****************************************************

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Some more unusual ones: http://www.muppetlabs.com/~breadbox/intercal-man/tonsila.html -----Original Message----- From: Gary Gapinski [mailto:Gary.Gapinski@grc.nasa.gov] Sent: 24 July 2003 14:37 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] the 'pound' and '#' are the same? (OT Rambling) On Thursday 24 July 2003 01:21, John Laur wrote: > I haven't ever

I don't have any experience of your problem - but I thought this might help. http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html <http://www.hut.fi/Misc/Electronics/circuits/uk_wiring.html> The UK (and some of it's former colonies) use a system called 3-wire ringing. Some equipment from overseas requires an adaptor to make it work. I don't know if the LineJack is one

I too am having this problem reported by Frank Hoonhout. Asterisk runs fine for a few minutes and then stops accepting new calls. (I have a standalone server with SIP phones and I'm not doing any external registration). Asterisk CVS-04/07/03-09:28:50 0x420e0037 in poll () from /lib/i686/libc.so.6 (gdb) info threads 16 Thread 14351 (LWP 7258) 0x420e187e in select () from