Cisco SIP problems and issues with Siemens VoIP switch

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Feb 27th, 2009
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Hi guys,

I hope someone can help, I've been trying to fix these issues for weeks now, and I don't see the solution 

The topology is like this:

Siemens HiPath 3500 v.7.0 switch with Siemens HG 1500 (VoIP module) connected on Cisco catalyst 3550 PoE switch with Cisco 7940 IP Phones using SIP protocol v.8.11.0.

Communication between IP Phones is established but here are the issues:

1. Assisted call transfer cannot be made. (Transfer is possible if transfer pressed and after number is dialed transfer pressed again, prior third number answers)

2. Blind call transfer does not work either (BlndXfr)

3. Call forwarding is not working as well (CFwdALL)

4. When call is made (between two IP phones locally) name of the caller is not presented, only the number of locale

5. No dial tone for free line out, only initial tone when you press speaker or pick up the handset

6. No ringing tone when outside line is dialed, if number dialed answers, connection is properly established. Proper busy tone when line is busy.

These are most critical issues since all of these functions are basic ones. I have tried transferring the call with digital Siemens set and it is working, so switch should work properly, but I don't know where to look for the solution, is it SIP is it Cisco IP phone or some (mis)configuration on the switch.

I would appreciate any solution or shared experience…

Thanks in advance,


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Nicholas Matthews Wed, 02/25/2009 - 15:02
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If this isn't CUCM/CME controlled you can disregard that command.

Once you're using a SIP phone with another call agent, there really isn't anyone on this forum that will be able to help you with signaling-level problems. That is something you would need to address on the proxy.

All of the SIP signaling options are going to be in the firmware, which we can't change. You can try downloading the most recent firmware to see if that helps, otherwise you'll need to work with whatever device you're controlling the phone with.


stojanandov Tue, 03/03/2009 - 06:05
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Hi Nick,

Thanks for the reply... I'm trying to locate the problem, but I cannot isolate it, is it in the phone (firmware), switch or proxy...or the combination of those...



Steffen.Baier Fri, 02/27/2009 - 08:16
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Have you tried to check in a wireshark trace to see if the Information is included in the SIP Header ?

Check a call from a Siemens Optipoint Phone to a Cisco SIP Phone and the other way around in order to establish if you have configured the Cisco Phone correctly.


stojanandov Mon, 03/02/2009 - 05:53
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Hi Steffen

Thanks for the suggestion, I've tried wireshark and capture packets from the network… I analyzed SIP ones and I get the following info:

At the beginning I get one packet with ICMP Port Unreachable , Packet 3,730 (PBX IP -> CISCO Phone IP)

When I set call forward on the phone (call made from line 100 to line 115 which is forwarded to 101), I get error “Reorder” on the phone, and this packet info:

SIP - Session Initiation Protocol

Response: SIP/2.0 302 Moved Temporarily ..

Via: SIP/2.0/UDP 10.x.x.x:5060;branch=z9hG4bKd5aec56921695ba26.6b7fc4251d0355680;rport ..

From: 100 ;tag=e0510f6b26 ..

To: ;tag=001d705ebfb9001a38cb6dc0-2813f05e ..

Call-ID: 8f7e6d7128f1d3d5 ..

CSeq: 8408 INVITE ..

Server: Cisco-CP7940G/8.0 ..

Contact: ..

Diversion: "115" ;reason=unconditional;privacy=off;screen=yes ..


Content-Length: 0 .... 567

For Call transfer, I get error message “transfer failed” and there are 3 packets:

INVITE (calling number to be transferred to)

200 OK (after second number responds)

ACK (after transfer is pressed and line dropped)

Can you suggest what to look for in these packets?


Nicholas Matthews Mon, 03/02/2009 - 10:38
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You may want to try this:

voice service voip

no supplement sip moved


stojanandov Tue, 03/03/2009 - 00:55
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Sorry for the question, it might sound stupid, but where to try those commands?


stojanandov Tue, 03/03/2009 - 05:29
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You mean in SIPConfig.cnf? I dont have any other config file...


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