Hi, all!

I have set up a Trixbox (2.8) server, but it accept calls only sometimes. It may be a ok for a little while (a very few minutes), but for a while (probably mostly) there is something wrong.

In the log, all incoming calls starts like this:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6

For calls that are accepted normally, the log continues normally. But if not, the logging ends here, and the caller finally gets a busy-signal (or something like that, probably generated by the voip-provider)…

Any ideas?

If it is of interest, the server is behind a firewall that has the ports udp 5060 and udp 10000-10100 forwarded to the server, rtp.sip changed accordingly, and externip/localnet is specified in sip_nat.conf. The firewall is a Juniper ssg5 (firmware Version: 6.1.0r3.0), with SIP and H323 disabled (as requested by the voip-provider).

Thanks in advance!

Frode

Hi,

I have 3 extensions (201, 202 & 203).
201 is X-Lite on my mac, 202 is siphon on my iPhone and 203 goes to a D-Link DVG-1120s which has a Philips voip433 connected to it.

When I call 201 -> 202, or 202 -> 201, or 203 -> 201/202 it all works. But when I want to call to extension 203 it doesn’t work. So I can call from 203 but not to 203.
Outbound calls go to an IVR, option 1 goes to call group 600 which is 201, 202 and 203. 201 and 202 ring, but not 203.

201, 202 and 203 have the same SIP settings. Even tried logging in as 201 using the D-link, but get the same problem but then for extension 201.

How can I solve this issue?

Best regards,
Paul Peelen

Hi, I’m kinda newb on trixbox and I’m stuck in a problem for days. I created an app on my company to find some info about the caller based on its number. Basically I send an http get:

http://callerid.intranet.tibra/retornacallerid.ashx?callerid=4196…

The website then returns the response in plain text:

D Paula Flores - cod 0 (Marli)

On trixbox, I set up the Caller ID Lookup Source like this:

Source description: http
Source type: HTTP
Cache results: no
Host: http://callerid.intranet.tibra
Port: 80
Path: /retornacallerid.ashx
Query: callerid=[NUMBER]

There’s no username or password set.

Then I related this source to my Inbound Routes. But still, when a call comes in, nothing happens, on sip debug no event raise concerning the cid lookup, looks like trixbox isn’t “trying” to get the cid like I configured.

My trixbox version is 2.6.

Am I missing something?

Hello,

I am just curius if this is normal Trixbox behaviour:

asterisk1*CLI> show channels
Channel Location State Application(Data)
Zap/1-1 s@from-pstn:1 Up Bridged Call(SIP/2137-b79169c0
SIP/2137-b79169c0 s@macro-dialout-trun Up Dial(ZAP/g0/18772344xxx|300|)
Zap/5-1 s@from-pstn:1 Up Bridged Call(SIP/2128-b720e208
SIP/2128-b720e208 s@macro-dialout-trun Up Dial(ZAP/g0/18669104xxx|300|)
4 active channels
2 active calls

Is it normal that if 2 calls is make 4 channels is occupying? I thoght one call = one channel. So if we use T1 = 23 channels only 11 calls can be made? It can’t be like that.
I assume I am wrong. Can someone explain?

Thank You,

MST

Hey all - I’m wondering if there is a way to have Asterisk bind to a second network port on the same Interface. Would it be possible to have two “bindport=xxxx” lines in sip.conf?

Thanks!
-Eq

I’m trying to set up a general delivery voicemail box for a few customer service reps. I have an SIP trunk from inphonex and all of the employees are using x-lite 3.0. The voicemail icon in x-lite needs to be functional and alert them when there is a voicemail. They *98 feature code + voicemail box# would work, but its too confusing and time consuming for them.

What direction do you think would be best? I don’t want to do email because there’s no way to delete or move messages to inbox via email. (and I can’t set up IMAP)

Any suggestions?

I’m not exactly sure what the problem here is. I’m getting a 404 on the DID Number it seems. I have my inbound route DID Number set to my SIP-ID, and set to an extension as my destination. I can place outbound calls just fine.

<— SIP read from 204.155.28.10:5060 —>
INVITE sip:MYSIPID@208.125.123.163 SIP/2.0
Record-Route:
Record-Route:
Record-Route:
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.20;branch=z9hG4bK5dfb9758
Via: SIP/2.0/UDP 204.155.29.58:5060;branch=z9hG4bK5dfb9758;rport=5060
Max-Forwards: 67
From: “4156398597″ ;tag=as50e8ba71
To:
Contact:
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-Timeout: 120
Content-Type: application/sdp
Content-Length: 383

v=0
o=root 336087564 336087565 IN IP4 204.155.29.58
s=sipgate VoIP GW
c=IN IP4 204.155.29.58
t=0 0
m=audio 19352 RTP/AVP 0 8 3 18 112 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<————->
— (19 headers 17 lines) —
Sending to 204.155.28.10 : 5060 (NAT)
Using INVITE request as basis request - 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
Found peer ‘SipGate’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 101
Peer audio RTP is at port 204.155.29.58:19352
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0×8 (alaw), peer - audio=0×90e (gsm|ulaw|alaw|g726|g729)/video=0×0 (nothing), combined - 0×8 (alaw)
Non-codec capabilities (dtmf): us - 0×1 (telephone-event), peer - 0×1 (telephone-event), combined - 0×1 (telephone-event)
Peer audio RTP is at port 204.155.29.58:19352
Looking for MYSIPID in ext_did (domain 208.125.123.163)
trixbox*CLI>
<— Reliably Transmitting (NAT) to 204.155.28.10:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0;received=204.155.28.10
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.20;branch=z9hG4bK5dfb9758
Via: SIP/2.0/UDP 204.155.29.58:5060;branch=z9hG4bK5dfb9758;rport=5060
From: “4156398597″ ;tag=as50e8ba71
To: ;tag=as168f3c9e
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<————>
Scheduling destruction of SIP dialog ‘319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com’ in 7424 ms (Method: INVITE)
trixbox*CLI>
<— SIP read from 204.155.28.10:5060 —>
ACK sip:MYSIPID@208.125.123.163 SIP/2.0
Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bK7fdf.a2434d06.0
Via: SIP/2.0/UDP 172.30.20.1;branch=z9hG4bK7fdf.a2434d06.0
From: “4156398597″ ;tag=as50e8ba71
Call-ID: 319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com
To: ;tag=as168f3c9e
CSeq: 103 ACK
Max-Forwards: 69
Content-Length: 0
X-hint: rr-enforced

<————->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘319761ae61e4ce6c7bc6d4512dfa3da1@sipgate.com’ Method: ACK
Really destroying SIP dialog ‘9e3ab88cf8ba88ce’ Method: REGISTER
Really destroying SIP dialog ‘e1e50191017e798e’ Method: REGISTER

Hi,

As a newbie I was happy with the way I was getting Trixbox up and running. All the searching and help from you guys made it a fairly easy install.

That is till I got a few Cisco 7941’s.
Firstly after searching I found that people said I needed to flash them with a SIP firmware.
So hastly I found pos3-07-5-00 and pos3-8-7-00.
Then I tried to use this guide : http://www.greenwireit.com/blog/2009/04/reflash-your-cisco-7940-7…

trouble is I just cannot get it to work.
My phone is now alternating between the cisco boot screen and the upgrading screen but no upgrade.

I have setup a laptop, switch and the phone on its own network . Fixed the ip on my laptop to 192.168.1.10 as per the links pictures but still no upgrade.
Can anyone offer any advise and help me through this.

DAve.

Does anyone know if I can pickup calls from a ZAP extension by using *8 on a SIP extension?
Call group and pickup group are the same, of course!

when the time is not met my calls should be routed to a IVR

== Manager ‘admin’ logged off from 127.0.0.1
— Executing [+15852003547@from-sip-external:1] NoOp(”SIP/4.68.250.148-b7d00468″, “Received incoming SIP connection from unknown peer to +15852003547″) in new stack
— Executing [+15852003547@from-sip-external:2] Set(”SIP/4.68.250.148-b7d00468″, “DID=+15852003547″) in new stack
— Executing [+15852003547@from-sip-external:3] Goto(”SIP/4.68.250.148-b7d00468″, “s|1″) in new stack
— Goto (from-sip-external,s,1)
— Executing [s@from-sip-external:1] GotoIf(”SIP/4.68.250.148-b7d00468″, “1?from-trunk|+15852003547|1″) in new stack
— Goto (from-trunk,+15852003547,1)
— Executing [+15852003547@from-trunk:1] NoOp(”SIP/4.68.250.148-b7d00468″, “Catch-All DID Match - Found +15852003547 - You probably want a DID for this.”) in new stack
— Executing [+15852003547@from-trunk:2] Goto(”SIP/4.68.250.148-b7d00468″, “ext-did|s|1″) in new stack
— Goto (ext-did,s,1)
— Executing [s@ext-did:1] Set(”SIP/4.68.250.148-b7d00468″, “__FROM_DID=s”) in new stack
— Executing [s@ext-did:2] Gosub(”SIP/4.68.250.148-b7d00468″, “app-blacklist-check|s|1″) in new stack
— Executing [s@app-blacklist-check:1] LookupBlacklist(”SIP/4.68.250.148-b7d00468″, “”) in new stack
— Executing [s@app-blacklist-check:2] GotoIf(”SIP/4.68.250.148-b7d00468″, “0?blacklisted”) in new stack
— Executing [s@app-blacklist-check:3] Return(”SIP/4.68.250.148-b7d00468″, “”) in new stack
— Executing [s@ext-did:3] GotoIf(”SIP/4.68.250.148-b7d00468″, “1 ?cidok”) in new stack
— Goto (ext-did,s,5)
— Executing [s@ext-did:5] NoOp(”SIP/4.68.250.148-b7d00468″, “CallerID is “WIRELESS CALLER” <+1585XXXXXXXXn>”) in new stack
— Executing [s@ext-did:6] Set(”SIP/4.68.250.148-b7d00468″, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
— Executing [s@ext-did:7] SetCallerPres(”SIP/4.68.250.148-b7d00468″, “allowed_not_screened”) in new stack
— Executing [s@ext-did:8] Goto(”SIP/4.68.250.148-b7d00468″, “app-daynight|0|1″) in new stack
— Goto (app-daynight,0,1)
— Executing [0@app-daynight:1] GotoIf(”SIP/4.68.250.148-b7d00468″, “0?”) in new stack
== Auto fallthrough, channel ‘SIP/4.68.250.148-b7d00468′ status is ‘UNKNOWN’

I am having problem getting my Cisco 7960G to connect to my GXE5028. My X-lite softphone works fine. I have all the name, password and extension setup in GXE to be the same (ex: all set to: 6001).  I am using SIP firmware P0S3-07-5-00 on the 7960. Please help.

With its new ClariLink family of WAN link controllers, Ecessa gives small and midsized enterprises the load balancing and failover capabilities for network traffic previously unavailable to small and midsized enterprises. The ClariLink devices let IT administrators deploy SIP traffic across multiple WAN or service provider networks, offering redundancy and automatic failover in the event that a VOIP session is interrupted for any reason. If a VOIP session is ended because of a network problem, the call is automatically sent to another link, with little to no interruption.
- Ecessa is rolling out a new family of WAN link controllers aimed at improving the performance and scalability of VOIP communications. The ClariLink WAN link controller, announced May 12, balances SIP (Session Initiation Protocol) traffic across multiple WAN links and service provider networks, en...

Hello to everyone,

i have buy 3 Grandstream BT-200, just to try it , after a new installation of Trixbox 2.2.4 ….. i create 3 new extensions an and i receive from asterisk console :

Name/username Host Dyn Nat ACL Port Status
104/104 192.168.0.94 D N 5060 OK (25 ms)
103/103 192.168.0.93 D N 5060 OK (26 ms)
102/102 192.168.0.90 D N 5060 OK (6 ms)

Example:
I make a call from 102 to 103, after pickup i want transfer this call to 104 …..
102 -> call -> 103
103 -> FLASH -> 104
after to complete i push TRANSFER
now 102 -> talk to -> 104

but i receive on display CALL ON HOLD and my BT-200 ring !!!! after pickup i see on screen 503 …. this is an server side error (503 Service unavailable)

I have another pc where i have installed asterisk 1.4 and freepbx 2.5.0 and this problem dont appear !!!!

PLS HELP ME !!!

TNX
SAT

—————————————————————————————————–

    -- Executing Macro("SIP/102-0859a568", "exten-vm|103|103") in new stack
    -- Executing Macro("SIP/102-0859a568", "user-callerid") in new stack
    -- Executing NoOp("SIP/102-0859a568", "user-callerid: device 102") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?report") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?start") in new stack
    -- Executing Set("SIP/102-0859a568", "REALCALLERIDNUM=102") in new stack
    -- Executing NoOp("SIP/102-0859a568", "REALCALLERIDNUM is 102") in new stack
    -- Executing Set("SIP/102-0859a568", "AMPUSER=102") in new stack
    -- Executing Set("SIP/102-0859a568", "AMPUSERCIDNAME=Commerciale") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?report") in new stack
    -- Executing Set("SIP/102-0859a568", "CALLERID(all)=Commerciale <102>") in new stack
    -- Executing Set("SIP/102-0859a568", "REALCALLERIDNUM=102") in new stack
    -- Executing NoOp("SIP/102-0859a568", "TTL:  ARG1: 103") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?continue") in new stack
    -- Executing Set("SIP/102-0859a568", "__TTL=64") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/102-0859a568", "Using CallerID "Commerciale" <102>") in new stack
    -- Executing Set("SIP/102-0859a568", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/102-0859a568", "VMBOX=103") in new stack
    -- Executing Set("SIP/102-0859a568", "EXTTOCALL=103") in new stack
    -- Executing Set("SIP/102-0859a568", "CFUEXT=") in new stack
    -- Executing Set("SIP/102-0859a568", "CFBEXT=") in new stack
    -- Executing Set("SIP/102-0859a568", "RT=15") in new stack
    -- Executing Macro("SIP/102-0859a568", "record-enable|103|IN") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/102-0859a568", "recordingcheck|20090415-124948|1239792588.12") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090415-124948|1239792588.12: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/102-0859a568", "No recording needed") in new stack
    -- Executing Macro("SIP/102-0859a568", "dial|15|tr|103") in new stack
    -- Executing DeadAGI("SIP/102-0859a568", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: priority is 1
  dialparties.agi: Caller ID name is 'Commerciale' number is '102'
  dialparties.agi: Methodology of ring is  'none'
       >  dialparties.agi: USE_CONFIRMATION:  'FALSE'
       >  dialparties.agi: RINGGROUP_INDEX:   ''
    --  dialparties.agi: Added extension 103 to extension map
    --  dialparties.agi: Extension 103 cf is disabled
    --  dialparties.agi: Extension 103 do not disturb is disabled
       >  dialparties.agi: extnum: 103
       >  dialparties.agi: exthascw: 1
       >  dialparties.agi: exthascfb: 0
       >  dialparties.agi: extcfb:
       >  dialparties.agi: exthascfu: 0
       >  dialparties.agi: extcfu:
    --  dialparties.agi: dbset CALLTRACE/103 to 102
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/102-0859a568", "SIP/103|15|tr") in new stack
    -- Called 103
    -- SIP/103-085a3e70 is ringing
    -- SIP/103-085a3e70 answered SIP/102-0859a568
    -- Music class default requested but no musiconhold loaded.
    -- Executing Macro("SIP/103-085b30f8", "exten-vm|104|104") in new stack
    -- Executing Macro("SIP/103-085b30f8", "user-callerid") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "user-callerid: device 103") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?report") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?start") in new stack
    -- Executing Set("SIP/103-085b30f8", "REALCALLERIDNUM=103") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "REALCALLERIDNUM is 103") in new stack
    -- Executing Set("SIP/103-085b30f8", "AMPUSER=103") in new stack
    -- Executing Set("SIP/103-085b30f8", "AMPUSERCIDNAME=INT Mirko") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?report") in new stack
    -- Executing Set("SIP/103-085b30f8", "CALLERID(all)=INT Mirko <103>") in new stack
    -- Executing Set("SIP/103-085b30f8", "REALCALLERIDNUM=103") in new stack
    -- Executing NoOp("SIP/103-085b30f8", "TTL:  ARG1: 104") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?continue") in new stack
    -- Executing Set("SIP/103-085b30f8", "__TTL=64") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,21)
    -- Executing NoOp("SIP/103-085b30f8", "Using CallerID "INT Mirko" <103>") in new stack
    -- Executing Set("SIP/103-085b30f8", "FROMCONTEXT=exten-vm") in new stack
    -- Executing Set("SIP/103-085b30f8", "VMBOX=104") in new stack
    -- Executing Set("SIP/103-085b30f8", "EXTTOCALL=104") in new stack
    -- Executing Set("SIP/103-085b30f8", "CFUEXT=") in new stack
    -- Executing Set("SIP/103-085b30f8", "CFBEXT=") in new stack
    -- Executing Set("SIP/103-085b30f8", "RT=15") in new stack
    -- Executing Macro("SIP/103-085b30f8", "record-enable|104|IN") in new stack
    -- Executing GotoIf("SIP/103-085b30f8", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing DeadAGI("SIP/103-085b30f8", "recordingcheck|20090415-124957|1239792597.14") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20090415-124957|1239792597.14: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/103-085b30f8", "No recording needed") in new stack
    -- Executing Macro("SIP/103-085b30f8", "dial|15|tr|104") in new stack
    -- Executing DeadAGI("SIP/103-085b30f8", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: priority is 1
  dialparties.agi: Caller ID name is 'INT Mirko' number is '103'
  dialparties.agi: Methodology of ring is  'none'
       >  dialparties.agi: USE_CONFIRMATION:  'FALSE'
       >  dialparties.agi: RINGGROUP_INDEX:   ''
    --  dialparties.agi: Added extension 104 to extension map
    --  dialparties.agi: Extension 104 cf is disabled
    --  dialparties.agi: Extension 104 do not disturb is disabled
       >  dialparties.agi: extnum: 104
       >  dialparties.agi: exthascw: 1
       >  dialparties.agi: exthascfb: 0
       >  dialparties.agi: extcfb:
       >  dialparties.agi: exthascfu: 0
       >  dialparties.agi: extcfu:
    --  dialparties.agi: dbset CALLTRACE/104 to 103
  == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing Dial("SIP/103-085b30f8", "SIP/104|15|tr") in new stack
    -- Called 104
    -- SIP/104-085bbbe8 is ringing
    -- SIP/104-085bbbe8 answered SIP/103-085b30f8
    -- Music class default requested but no musiconhold loaded.
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>'
    -- Executing Macro("SIP/103-085b30f8<ZOMBIE>", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/103-085b30f8<ZOMBIE>", "w") in new stack
    -- Executing NoCDR("SIP/103-085b30f8<ZOMBIE>", "") in new stack
    -- Executing GotoIf("SIP/103-085b30f8<ZOMBIE>", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/103-085b30f8<ZOMBIE>", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/103-085b30f8<ZOMBIE>", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/103-085b30f8<ZOMBIE>'
    -- Incoming call: Got SIP response 400 "Bad Request" back from 192.168.0.93
    -- Music class default requested but no musiconhold loaded.
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568' in macro 'dial'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568' in macro 'exten-vm'
  == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/102-0859a568'
    -- Executing Macro("SIP/102-0859a568", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/102-0859a568", "w") in new stack
    -- Executing NoCDR("SIP/102-0859a568", "") in new stack
    -- Executing GotoIf("SIP/102-0859a568", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing GotoIf("SIP/102-0859a568", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing Wait("SIP/102-0859a568", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/102-0859a568' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/102-0859a568'

I have 2 systems that are trunked together over IAX.

Box A has all the outside lines, Box B is a satellite location that dials out through Box A over the IAX trunk.

When someone dials from an extension on Box B, the call is routed through Box A and the call is made - GREAT! My problem is that for 911 and callback reasons, I need to be able to set the caller ID that is sent out (and yes, my provider allows for outbound CID overriding). I know how to do it if the call was made from a SIP extension (in the Outbound CID field), but how do I do the same thing for a call made from the IAX trunk?

Thanks,
dj