Hi. I just installed Elastix and can not seem to get a 2nd sip trunk working. A little background. We have been running a Trixbox server for several years with a Vitelity sip trunk. We then tried to add a 2nd sip trunk from XCast but when we dial out we get "All circuits are busy, please try your call again later" when attempting to make an outbound call (thru the XCast Trunk) The Vitelity trunk worked fine. In the process of working with XCAst tech support they recommended Elastix. So I installed Elastix on a test box and have the same issue. On the Elastix box I added the sip trunk providers in reverse order from how they were set up in Trixbox. I 1st added the XCast trunk and it works perfectly. I thought I was home free - but now after adding the Vitelity sip trunk when trying to make an outbound call thru the Vitelity sip trunk it now gets, "All circuits are busy......" The XCast sip trunk works fine. Same issue with both distributions. Seems to be a Asterisk issue, unless I am doing something wrong??? Here is the call log from attempting an outgoing call thru the 2nd added sip trunk with Vitelity.

[Oct 4 17:06:47] VERBOSE[25261] pbx.c: -- Executing [630xxxxxxx @Vitel-Outbound:5] Macro("SIP/358-00000012", "outisbusy,") in new stack
[Oct 4 17:06:47] VERBOSE[25261] pbx.c: -- Executing [s@macro-outisbusy:1] Progress("SIP/358-00000012", "") in new stack
[Oct 4 17:06:47] VERBOSE[25261] pbx.c: -- Executing [s@macro-outisbusy:2] GotoIf("SIP/358-00000012", "0?emergency,1") in new stack
[Oct 4 17:06:47] VERBOSE[25261] pbx.c: -- Executing [s@macro-outisbusy:3] GotoIf("SIP/358-00000012", "0?intracompany,1") in new stack
[Oct 4 17:06:47] VERBOSE[25261] pbx.c: -- Executing [s@macro-outisbusy:4] Playback("SIP/358-00000012", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[Oct 4 17:06:48] VERBOSE[25261] file.c: -- <SIP/358-00000012> Playing 'all-circuits-busy-now.gsm' (language 'en')
[Oct 4 17:06:49] VERBOSE[25261] file.c: -- <SIP/358-00000012> Playing 'pls-try-call-later.gsm' (language 'en')
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-outisbusy:5] Congestion("SIP/358-00000012", "20") in new stack
[Oct 4 17:06:52] WARNING[25261] channel.c: Prodding channel 'SIP/358-00000012' failed
[Oct 4 17:06:52] VERBOSE[25261] app_macro.c: == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/358-00000012' in macro 'outisbusy'
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: == Spawn extension (Vitel-Outbound, 630xxxxxxx, 5) exited non-zero on 'SIP/358-00000012'
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [h@Vitel-Outbound:1] Macro("SIP/358-00000012", "hangupcall,") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/358-00000012", "1?noautomon") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Goto (macro-hangupcall,s,3)
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:3] NoOp("SIP/358-00000012", "TOUCH_MONITOR_OUTPUT=") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/358-00000012", "1?skiprg") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Goto (macro-hangupcall,s,7)
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/358-00000012", "1?skipblkvm") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Goto (macro-hangupcall,s,10)
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/358-00000012", "1?theend") in new stack
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Goto (macro-hangupcall,s,12)
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: -- Executing [s@macro-hangupcall:12] Hangup("SIP/358-00000012", "") in new stack
[Oct 4 17:06:52] VERBOSE[25261] app_macro.c: == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/358-00000012' in macro 'hangupcall'
[Oct 4 17:06:52] VERBOSE[25261] pbx.c: == Spawn extension (Vitel-Outbound, h, 1) exited non-zero on 'SIP/358-00000012'

Just a note, we do have the Custom Context installed.

Any incite would be greatly appreciated.
-Rich

Rich,

Have you configured the trunks one by one in outbound routes on elastix?

Yes. 1st created the trunk then outbound route. Repeated for 2nd trunk with different provider. Then created a custom context for each outbound route. I see that the 2nd trunk added vitel-inbound & vitel-outbound shows a status of Unmonitored???

Name/username Host Dyn Nat ACL Port Status
358/358 192.168.0.0 D A 5060 OK (7 ms)
XCast-Inbound 38.102.250.60 5060 OK (78 ms)
XCast-Trunk 38.102.250.50 5060 OK (77 ms)
vitel-inbound/xxxxx 66.241.96.96 N 5060 Unmonitored
vitel-outbound/xxxxx 64.2.142.17 N 5060 Unmonitored
5 sip peers [Monitored: 3 online, 0 offline Unmonitored: 2 online, 0 offline]

-Rich

Sorry, I realize I did not answer your question. I do not have any inbound routes set up on my test Elastix box as I am only testing outbound at this time.

-Rich

Possibly this thread may shed some light, www.elastix.org/index.php/en/component/k...k-on-elastix-20.html . User rnrover seems to have a similar setup as yours.

What version of Elastix are you running? And can you post your trunk settings, minus any usernames/passwords.

Also your thread should probably be in the Help forum and not Announcements.


-John

Have you set the outbound callerID on the trunk configuration?
Also on the trunk settings add the following flag:
qualify=yes
And double check if you need or not a register string. If so, what is the output for:
asterisk -rx "sip show peers"
asterisk -rx "sip show registry"