Asterisk and Siemens HiPath 3800 / HG1500

Hello everybody,

I hoope that I am not alone in this world with such problems. I trying to connect HiPath 3800 / HG1500 and Asterisk but no luck for now. Does anybody have some experience and procedure how to do it? My software version on Siemens HG1500 is V8 and I use AsteriskNOW GUI 1.7.1 - 64bit.

Any advice will be welcome.
Thank you

Interesting…nobody know ???


How are you connecting the PBX-es? Via SIP/H323 or via Analog/BRI/BRI. This is more or less the basic information that we need to answer you :wink:

I apologize for late response because I’m very busy.
I would like to connect Siemens HiPath 3800 and Asterisk via SIP (on first place) or H.323. If work with SIP that would be great.

dejanst, I hope you can help me :exclamation: :question:


I do not have any experience with Siemens, but have lot’s of experience with Asterisk. I will try my best to help with solving this problem.

What are your settings for the SIP Trunk (for connecting Siemens and Asterisk) on Asterisk box? A copy/paste of the appropriate part of sip.conf would do just fine.

OK nice to hear. I am familiar with Siemens PBX . Regarding I don’t know which part is the appropriate part of sip.conf I can give you all . It doesn’t mater because I have this PBX in laboratoy and data is not confidentional.
I rename the file in .txt.

;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Wed Dec 1 10:35:32 2010
; SIP Configuration example for Asterisk
; SIP dial strings

;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;host= ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a “friend”
; so there’s currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;mailbox=1234@default ; mailbox 1234 in voicemail context “default”
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information

; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;regexten=1234 ; When they register, create extension 1234
;callerid=“Jane Smith” <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;directmedia=no ; Typically set to NO if behind NAT
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes ; Send a 100 Trying when the device registers.

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip= ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to “asterisk”
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultuser=polly ; Username to use in INVITE until peer registers
; Normally you do NOT need to set this parameter
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don’t work properly with “never”

;insecure=port ; Allow matching of peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it’s 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
; Call group and Pickup group should be in the range from 0 to 63
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip= ; IP address to use if peer has not registered
;deny= ; ACL: Control access to this account based on IP address
;permit= ; we can also use CIDR notation for subnet masks

;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip= ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.

;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.


Did you define a SIP Trunk in the WebGUI? I do not see it in the .conf file. There are tons of internet sites where they describe this. This is one excellent guide for Trixbox that is using FreePBX as a WebGUI; … _tears.pdf

Yes, I was define through WEB. I will check recomended pdf and I hope I will find proper information.
Will be in touch.


Did you apply the settings after you created the SIP Trunk in the WebGUI? Perhaps that is your problem :smile:

Yes, I appyl settings and nothing happend …till today.
In WEB browser I can not see trunk status, but when I try to make call from Siemens to Asterisk I find zhis lines in Asterisk log:

Dec 10 09:09:32] NOTICE[2833] chan_sip.c: Call from ‘trunk_1’ to extension ‘6300’ rejected because extension not found in context ‘DID_trunk_1’

If I undarstand trunk is alive?! I think I must find some rules for DID ?
Any advice in this direction ?


Please tell me output of the command “sip show peers” in Asterisk CLI.

dejanst, I put this comand in to CLI but return was:
-bash: sip: command not found

Can you tell me where to insert this commands.


To issue Asterisk CLI commands, you have to enter the Asterisk CLI.

Here is a Wiki page on connecting to the Asterisk CLI: … to+the+CLI

Malcomd thank you for sugested link. I already made SSH connection but command which dejanst sugest don’t work. Didi I mentioned I work with Asterisk NOW. I think there should be no diferent?


Did you perform “sip show peers” from the Asterisk CLI?


asterisk -r
(asterisk CLI loads)
sip show peers


Yes I did and finaly have result:
I apologizr, didn’t know I have to write command asterisk -r.
Ok What I see is:
localhostCLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
6001/6001 (Unspecified) D N 5060 Unmonitored
6002/6002 (Unspecified) D N 5060 Unmonitored
6003/6003 (Unspecified) D N 5060 Unmonitored
6004/6004 (Unspecified) D N 5060 Unmonitored
6005/6005 D N 5060 Unmonitored
6010/6010 (Unspecified) D N 5060 Unmonitored
siemens 5060 Unmonitored
7 sip peers [Monitored: 0 online, 0 offline Unmonitored: 7 online, 0 offline]

What is next step?


I am in same situation, trouble connecting the two… did You managed to get any results on this?