Siemens Hipath HG1500 to Asterisk FreePBX

This info came from Achmad Taufik who was kind enough to share the setup, should make it alot easier for anyone integrating the siemens hipath HG1500 with Asterisk.

If you want to do the same integration using PRI TSM2 card instead of SIP see http://joekane.eu/siemens-hipath-3750-to-asterisk-te110p

The HG1500 is an expansion card for the Hipath that enables SIP extensions and in this case Trunks.

This it the System Config

Hipath Manager E + Asterisk

Once you installed the hg1500 card, go the web based configuration panel. Create a node for the asterisk pbx. Explorers -> Voice Gateways -> Nodes

Hipath Manager E + Asterisk

Create call address for the node. All my asterisk extension start with digit 6XX.

Hipath Manager E + Asterisk

On the asterisk box, create sip trunk, the host=10.0.1.7 is the IP address of hipath hg1500 card.

Hipath Manager E + Asterisk

This the outbound route from asterisk to the hipath. It routes asterisk users dialing [1238]XX to the hipath SIP trunk.

Hipath Manager E + Asterisk

On the hipath create a dialplan for asterisk extensions.

Hipath Manager E + Asterisk

I had to create a virtual port on the hipath for all asterisk extensions (all extensions 6XX are virtual ports). Whenever I dial this extension on the hipath it forwards the calls to the SIP trunk. I did try using the dial plan without creating all virtual stations but it didn’t work.

 

Hipath Manager E + Asterisk

This is how I create a virtual port (see check sign on “Virtual Station”). Forward all calls to this station to SIP trunk.

Thats it, Thanks again to Achmad Taufik for the guide and screenshots. Any issues post a comment and we can try to fix.

Some extra screenshots for the Route config

24 thoughts on “Siemens Hipath HG1500 to Asterisk FreePBX

  1. I desperately need help: I followed the steps, but my Asterisk cannot register a trunk with HG1500, even though the trunk is online. May I ask what did you put into User Context and Register String in Asterisk SIP trunk configuration?I made some workaround: I created a SIP client on HG1500, and then trunk registers when I put this client number in Context and Register String, but calls from SIP phone to HP PBX extentions are not coming through, in HG1500 logs I can see only( TransmissionMgr "03/25/2010 09:50:21.525628" CSipPacketLog.cpp 199)<<OUT>> to 10.64.26.11:5060:SIP/2.0 487 Request Terminatedwhen making a call attempt.Any ideas are very much appreciated. Thank you.Denis.

  2. Hi Denis, I suspect there is no register string needed but I cant say for certain, this config was shared to me by Achmad I will ask him to take a look. Cheers, Joe

  3. Hi Joseph,I finally guessed what should have been done. The registry string must contain the Call no. of one of the extensions registered with the PBX as virtual ports. The solution you posted is great, the only drawback I can see is that every virtual station you create for an asterisk extenstion requires a licence as if you attach a new phone to HiPath. Can you confirm this please? Thanks very much to you and Achmad, you saved me a huge amount of time.Cheers,Denis.

  4. Hi Joseph,After some more tests in turned out that calls from SIP phones to HiPath extensions still not working.I think the only thing I’m missing to reproduce your configuration completely is a screenshot for TCP/IP route configuration. In the Dial plan screenshot in your post there are 2 tables, the lower shows TCP/IP route selected for route table no 6. Can you please post that screenshot?Thank you.Denis

  5. Hi Denis,I emailed Achmad to have a look, he should be able to answer your question, If I get any updates i’ll send them over.Cheers, Joe

  6. hi Joe and Denissorry for not checking this website and my email for quite sometime.as for Denis question,If SIP —> Hipath is not working I don’t think that "TCP/IP" route is the problem. That route is meant only for forwarding from hipath to asterisk.As long as the extension you dial (from the SIP) is available in Hipath as a virtual extension that should "ring". I advise to look in the setup of virtual port/station and the asterisk sip.conf.That TCP/IP route is setup automatically when you install hg1500 card. I can not remember doing anything special on that route.Anyway I am sending some additional screenshot to Joe’s email in case you really need further details on that route.And I don’t think you need register string for trunking with asterisk.Have you created an extensions in the asterisk that match the virtual station in hipath?keep me updated on the result.ThanksAchmad

  7. Denis,I forgot one thing, on the Manager E — Systemview choose call forwarding, setup a targetto dial a virtual station, in my case it was "65XXX" as my dial plan specifiy 65CZZ as TCP/IP route. "XXX" is the number of virtual station.You have to create one forward definiton for each virtual station.(this is not elegant way, but i could not find any other way) Then go to Station view, On each virtual port, set forwarding for Day, Night and Internal to match your forwarding table you set in the System view.This is my last step that I forgot to mention. I wish you sucess.Again, let me know what is the outcome

  8. Achmad,Finally, I got my system working. I pretty sure you could have yours working without creating extensions on HiPath for Asterisk clients. What software/firmware version your HG1500 card is running on?Cheers,Denis

  9. Joe,That’s the point, I didn’t do anything extra, I even didn’t create extensions. The key thing was the HG1500 software version, it must be HI-G15.75.004.S. After it is installed, everything starts working.Cheers,Denis.

  10. Hi Joseph,i’m missing two steps. You created a Route and called it TCP/IP but what did you configure for this route? And the other thing is the dial rule called 3 HG1500 SIP. What are the preferences for this? Thank youJohn

  11. Hi Joe,this helped me very much. I had a problem by making calls towards the HiPath but i fixed that in the sip.conf by setting up canreinvite=no. But i can not make a call from HiPath to Asterisk. Its ringing but asterisk does not get anything. sip set debug does not show any information. Is it possible that the HiPath is using TCP instead of UDP? You already helped me very much! Thank you again!John

  12. Hi Joe,everything is working fine now. i have deleted the virual ports on the hipath.just one problem is remaining. i can not call with a sip client to number outside the company

  13. Hello Joe,I’m trying to go through the same setup and wondering if you know of any other requirements to get this to work such as cornet licenses… From the asterisk side, it looks like the SIP trunk is active, but calls are not going through from Siemens to Asterisk and vice versa. I’m new to this Siemen system, and trying to figure out how to debug from the Siemens side. Any help would be greatly appreciated.Thanks much,Anthony

  14. Not sure if John still reads this post, but in order to call outside numbers from Asterisk sip client, the outbound route needs to be configured to send all calls through the asterisk-hg1500 sip trunk. Say the user has to dial 9 first, then there will be an entry: 9XXXXXXXXXX in the outbound route.The issue that I am having now is I can not have the same virtual station (say 8001) on the hipath to be forwarded to extension 8001 on Asterisk (no tone). Wondering if Joe or anyone who is following this post can help.Thanks,Anthony

  15. Hi Anthony, I wont have access to a system until Monday – Im using PRI to connect to asterisk so the setup is a little different – I’ll email Achmad.

  16. Thanks Joe, looking forward to hearing from you. Everything seems to be working for me now using a SIP trunk, I’ll eventually switching over to a PRI trunk due to licensing restriction (having a 2 channel sip trunk does not allow me to have more than 2 calls at the same time). Another thing that I’m trying to work on is the CID issue making sure the extensions show up correctly when calls are made internally between extensions but calls to have the outbound CID when going through the SIP trunk to the Siemens box and to the outside.Anthony

  17. Hi, I still cannot figure out what you setup for the Dial rule " HG1500 SIP" on your HiPath.How to define this rule? I cannot find it on your screenshots, unfortunately.

  18. we have siemens HiPath 3800 PBX with STMI2 card alredy it have one sip trunk line can u give me advise how to connect this system with your asterrisk soft PBX i read ur document and try to do it but it is not working i want to know how can i get sip trunk line from asterisk PBX

  19. Good evening, I’m having trouble performing the interlinking, I can call to the asterisk x siemens but not siemens for asterisk, could you help me, I can not download extra images the link is broken. Thank you.

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