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...{HANGUPCAUSE}|1; Hangup(); }; But the goto dialresult is not executed: Executing [from-sip:1] Macro("SIP/7XX-b403", "DialExternal|xxxxxx") in new stack -- Executing [macro-DialExternal:1] Dial("SIP/7XX-b403", "Zap/G3/07803034440|120|g|M(connected^jmls@mike^706)") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G3/XXXXXX -- Zap/10-1 is proceeding passing it to SIP/7XX-b403 -- Zap/10-1 is ringing -- Hungup 'Zap/10-1' == Spawn extension (macro-DialExternal, s, 1) exited non-zero on ...

...${EXTEN:3}) exten => _4427XX,3,Hangup() exten => fax,1,Goto(fax,1,1) [macro-faxreceive] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten => s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN}) exten => s,3,rxfax(${FAXFILE}) exten => s,103,SetVar(EMAILADDR=jmls@tessera.co.uk) exten => s,104,Goto(3) [fax] exten => 1,1,Macro(faxreceive) exten => h,1,system(/usr/sbin/mailfax ${FAXFILE} ${EMAILADDR} \ "${CALLERIDNUM} ${CALLERIDNAME}") exten => h,2,Hangup()

...y tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Fix a crash in res_jabber by ensuring that we don't alter memory after it's freed. (Closes issue #17387. Reported, tested by jmls. Patched by tilghman) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Teste...

...isk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix) (Closes issue #18951. Reported by jmls. Patched by wdoekes) * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue. This issue was found and reported by the Asterisk test suite. (Closes issue #18951. Patched by mnicholson) * Resolve potential crash when using SIP TLS support. (Closes issue #19192. Report...

...y tbelder) * Ensure user portion of SIP URI matches dialplan when using encoded characters (Closes issue #17892. Reported by wdoekes. Patched by jpeeler) * Fix a crash in res_jabber by ensuring that we don't alter memory after it's freed. (Closes issue #17387. Reported, tested by jmls. Patched by tilghman) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Teste...

...isk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix) (Closes issue #18951. Reported by jmls. Patched by wdoekes) * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue. This issue was found and reported by the Asterisk test suite. (Closes issue #18951. Patched by mnicholson) * Resolve potential crash when using SIP TLS support. (Closes issue #19192. Report...

...by Chris Hillman) * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets (Reported by Thava Iyer) * ASTERISK-23233 - alembic missing scripts for certain realtime tables (Reported by jmls) * ASTERISK-22537 - Create Sorcery equivalent to the AST_CONFIG function (Reported by George Joseph) * ASTERISK-23120 - ARI/AMI: allow objects created via interfaces to have their unique ID specified by the external application (Reported by Matt Jordan) * ASTERISK-22008 - Confi...

...by Chris Hillman) * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets (Reported by Thava Iyer) * ASTERISK-23233 - alembic missing scripts for certain realtime tables (Reported by jmls) * ASTERISK-22537 - Create Sorcery equivalent to the AST_CONFIG function (Reported by George Joseph) * ASTERISK-23120 - ARI/AMI: allow objects created via interfaces to have their unique ID specified by the external application (Reported by Matt Jordan) * ASTERISK-22008 - Confi...

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots