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Externally-accessible conference?

davidm870's picture
Posted January 24th, 2008 by davidm870 (tadpole)

Good $GREETING_TIME

We'd like to set up a conference that's accessible via one of our PSTN DIDs. Anybody know how we might do that?

Thanks in advance,

David Mitchell


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route the DID to the

p_lindheimer's picture
On January 25th, 2008 p_lindheimer (Developer) said:

route the DID to the conference or to an IVR with the conference as an option if you need to get fancier.


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If the conference doesn't

jono's picture
On January 29th, 2008 jono (tadpole) said:

If the conference doesn't appear as a destination choice of a DID in Inbound Routing (older versions didn't include conferences as a destination), simply create a ring group with the conf number followed by a # as the extension to ring. Then point your DID at the ring group.


Conference work now, thanks, but PIN doesn't

davidm870's picture
On February 7th, 2008 davidm870 (tadpole) said:

External conference work after struggling with a bit of ambiguity (putting 206XXXYYYY in the DID Number field didn't work, but putting YYYY there did). However, since I don't want the whole world using our conference number I added a user PIN to the conference configuration. Now calls from either inside or outside to the PINned conference number get terminated almost instantaneously with the following in the CLI:

-- Executing Set("SIP/4321-08330b48", "CALLERID(all)="David Mitchell" <4321>") in new stack
-- Executing Set("SIP/4321-08330b48", "REALCALLERIDNUM=4321") in new stack
-- Executing NoOp("SIP/4321-08330b48", "TTL: ARG1: ") in new stack
-- Executing GotoIf("SIP/4321-08330b48", "0?continue") in new stack
-- Executing Set("SIP/4321-08330b48", "__TTL=64") in new stack
-- Executing GotoIf("SIP/4321-08330b48", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("SIP/4321-08330b48", "Using CallerID "David Mitchell" <4321>") in new stack
-- Executing Set("SIP/4321-08330b48", "MEETME_ROOMNUM=6769") in new stack
-- Executing GotoIf("SIP/4321-08330b48", "0?READPIN") in new stack
-- Executing Answer("SIP/4321-08330b48", "") in new stack
-- Executing Wait("SIP/4321-08330b48", "1") in new stack
-- Executing Set("SIP/4321-08330b48", "PINCOUNT=0") in new stack
-- Executing Read("SIP/4321-08330b48", "PIN|enter-conf-pin-number||||") in new stack
-- User disconnected
== Spawn extension (from-internal, 6769, 7) exited non-zero on 'SIP/4321-08330b48'
-- Executing Macro("SIP/4321-08330b48", "hangupcall") in new stack
-- Executing ResetCDR("SIP/4321-08330b48", "w") in new stack
-- Executing NoCDR("SIP/4321-08330b48", "") in new stack
-- Executing GotoIf("SIP/4321-08330b48", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/4321-08330b48", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing GotoIf("SIP/4321-08330b48", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing Hangup("SIP/4321-08330b48", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/4321-08330b48' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/4321-08330b48'

We're using FreePBX 2.3.1.3 over Asterisk SVN-branch-1.2-r82334

Thanks!


Pin problem on concerences

justincase's picture
On February 21st, 2008 justincase (tadpole) said:

Also using 2.3.1.3 here. I see the following in logs when I try to us a PIN protected conference:

[Feb 21 11:39:20] VERBOSE[4727] logger.c: -- Executing [5234@from-internal:7] Read("SIP/167-ac04aba0", "PIN|enter-conf-pin-number||||") in new stack
[Feb 21 11:39:20] WARNING[4727] file.c: File enter-conf-pin-number does not exist in any format
[Feb 21 11:39:20] WARNING[4727] file.c: Unable to open enter-conf-pin-number (format 0x4 (ulaw)): No such file or directory
[Feb 21 11:39:20] VERBOSE[4727] logger.c: -- User disconnected

It seems your problem happens at the same spot. I have confirmed I don't have the file enter-conf-pin-number but I do have others in /var/lib/asterisk/sounds/ such as conf-getpin.gsm

Any idea where the properly named sound files are?

I see now that Freepbx 2.4 is out and stable. Last I had looked it was still alpha. Perhaps upgrading will fix this. I'll let you know how it goes!


Thanks for the hint, justincase. It works now

davidm870's picture
On February 21st, 2008 davidm870 (tadpole) said:

Thanks for your better verbose logging. I copied /var/lib/asterisk/sounds/conf-getpin.gsm to /var/lib/asterisk/sounds/enter-conf-pin-number.gsm and Allison now asks for a PIN and gives us a conference. No doubt all will be better in 2.4 in any case.

David Mitchell


enter-conf-pin-number not found

jpforte's picture
On October 1st, 2008 jpforte (tadpole) said:

I just upgraded from 2.4 to 2.5 and got the same problem! I copied the files as mentioned in last post and that fixed it. Did I upgrade wrong?


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