Re: [asterisk-users] H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi shinacaly...@gmail.comwrote: Hi! I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide a conference bridge for an existing Avaya PBX. I have no control over the Avaya system, but I am able to speak with the admin in charge when I need stuff done. I am running all this in a VirtualBox virtual instance, with CentOS 5.4 as the asterisk's host operating system. I configured a h323 trunk asterisk based on a few guides I discovered online, and I created a single sip extension (to test), and I am able to make a call from the Avaya PBX extensions successfully to my asterisk-freepbx virtual machine. The problem is when I try to make calls from Asterisk to Avaya, I get no sound whatsover and the call just keeps trying indefinitely until I end it. (I've used Twinkle and Ekiga softphones). This is what I find in the logs: [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/16000-0002, 0|AGI|fixlocalprefix) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:13] Set(SIP/16000-0002, OUTNUM=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:14] Set(SIP/16000-0002, custom=AMP) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:15] ExecIf(SIP/16000-0002, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/16000-0002, dialout-trunk-predial-hook|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/16000-0002, ) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/16000-0002, 0?bypass|1) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/16000-0002, 1?customtrunk) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,21) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:21] Set(SIP/16000-0002, pre_num=AMP:h323/Avaya/) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:22] Set(SIP/16000-0002, the_num=OUTNUM) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:23] Set(SIP/16000-0002, post_num=@ 10.100.7.15:1720) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:24] GotoIf(SIP/16000-0002, 1?outnum:skipoutnum) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,25) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:25] Set(SIP/16000-0002, the_num=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:26] Dial(SIP/16000-0002, h323/Avaya/18...@10.100.7.15:1720|300|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Requested transfer capability: 0x00 - SPEECH my h323.conf file is below: [general] port = 1720 bindaddr = 10.101.4.224 amaflags = AVAYA progress_setup = 8 progress_alert = 8 faststart = yes h245tunneling = yes gatekeeper = DISABLE disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 context=from-internal h323id=ObjSysAsterisk callerid=testbridge logfile=/var/log/asterisk/h323_log [Avaya] type=friend context=from-internal host=10.100.7.15 port=1720 disallow=all allow=g729 allow=g723 canreinvite=no dtmfmode=rfc2833 Please help me find out why the call isn't going through. -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690 -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690
[asterisk-users] DUNDi Confusion
Dear community,
Please help. I've been looking around the internet (and in this great forum)
for help with DUNDi setup between servers (I'm using Elastix) and while I
can get my servers to lookup extensions on each other very well, I have not
been able to successfully make calls between servers. For my test
environment, I have 3 servers setup for now, and these are the steps I've
followed:
1. I edited dundi.conf on each server to have the following info:
(this listing is for all servers)
[mappings]
priv = ext-dundi,0,IAX2,priv:${SECRET}@
'server-hostname'/${NUMBER},nopartial
[00:1C:C0:65:34:04]
model = symmetric
host = 192.168.1.128
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
[08:00:27:57:6E:0E]
model = symmetric
host = elastix-1
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
[08:00:27:15:0E:F1]
model = symmetric
host = elastix-2
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
2. I also edited extensions_custom.conf in each server to have:
[ext-dundi]
include = ext-local
include = ext-paging
include = ext-intercom-users
include = ext-group
include = ext-meetme
3. I also created an IAX2 Trunk called 'priv' using FreePBX (placing
information below only within the PEER Details(this trunk shows up as
'IAX2/priv' in FreePBX/Elastix web configurator):
[priv]
type=friend
dbsecret=dundi/secret
context=from-internal
trunk=yes
4. I also created a DUNDi Trunk called 'priv' as well in FreePBX and edited
only the DUNDI Mapping in there. This too shows up as 'DUNDi/priv' in the
FreePBX/Elastix web configurator.
The next steps to do is what confuses me. My DUNDi lookups and queries work
fine, and I have no firewalls between the boxes.
I have created a route called dundi-outside in each server's FreePBX that
references the DUNDi/priv route, and subsequently deleted it, because
whenever i try to make calls i get either an 'all-circuits-are-busy' error
msg, or i get a
'call-cannot-be-completed-as-dialled-please-check-the-number-and-dial-again'
error.
I'm really confused as what is going wrong. Am I (surely) missing something?
Any help will be greatly appreciated.
Hope to hear from you soon.
--
best regards,
Sina Owolabi
2348034022578
23417203257
23417420690
--
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[asterisk-users] AsteriskNOW + legacy PBX integration
Hi, I wonder if this question has been answered before, but im kind of stuck.. I have been trying to setup AsteriskNOW with a Digium TDM844B card with 4FXS/4FXO modules.. trying to connect it with a Panasonic KT616 PABX.. this has 6CO ports and 16 extensions. All the extensions are used up, the only free ports are the CO ports which have never been used. My layout is to connect PSTN connections to the 4FXO ports , and have the 4FXS ports connect to the Panasonic PABX. I wish to be able to have asteriskNOW as the telephony gateway to the organization, from the PSTN lines. There is a remote office with about 5 users, i expect to be able to have them use SIP phones, as the two offices are connected with a high bandwidth radio connection. I wish to be able to use asteriskNOW for interoffice calling, IVR, and call hunting. My confusion is how to setup the Panasonic PABX on asteriskNOW.. so that SIP users can dial extensions on the Panasonic PABX, and the Panasonic extensions can dial the SIP users in the remote office on AsteriskNOW. How do i properly define the Panasonic,in asteriskNOW, so that this is possible? Without breaking any configs? I tried adding new contexts to the extensions.conf but they were not recognized.. How do i properly edit the existing dial plan to include my needs? I also need to be able to achieve this fairly graphically so that if i need to leave, the other designated IT guys or a member of staff can make changes to the system without messing with configuration files, and it would be a lot easier to support. Im trying to convince the boss to invest in a channel bank, or astribank, but im not having much luck as i have to justify the TDM844B or its my salary :) Can anyone advise? Thanks very much in advance -- Shina Owolabi 2348034022578 23417203257 2341360480 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
