External SIP calls to/from local SCCP Phones

[spring329]

Hi,

I have a 2901 ISR with two 6911 Phones connected. I have set up the router config and have working calls between the two phones.

I have now connected a laptop to the router and have installed Linphone soft phone. I would now like to make calls between Linphone and the 6911 phones. I have configured the router and have made some progress. I am able to initiate calls between the two, however when I pick up the call it is immediately terminated.

My router outputs the following error, suggesting it is a codec issue:

%DSMP-3-DSPALARM: Alarm on DSP : status=0x0 message=0x0 text=N

Looking at Linphone settings, it seems there is a common codec "G722", which I have enabled. I have a PVDM3 in the router and have set up a dspfarm and specified the g722-64 codec. I have then specified the same codec in the dial-peer. I am not sure if a dspfarm is correct way to go about configuring a simple setup like this, or if I have configured it correctly. I'm also not sure which of 48/56/64kbps for G722 is supported by Linphone.

I have attached my router config (some lines removed) and a image of my (simple!) network setup.

Can anyone spot what might be causing the issue with the instant termination. I would also appreciate any comment on the configuration, particularly if there are voip-related settings that are in there needlessly.

Many thanks

[Aditya Gupta]

Hi

Can you attached debug ccsip messages output from the router when attempting to establish call between linphone and 6911

Regards

Aditya Gupta

[spring329]

Hi,

Two outputs attached (call originated from each point)

Also here is the output of ccsip error:

router1#debug ccsip error
SIP Call error tracing is enabled
router1#
SIP: (209) Attribute mid, level 1 instance 1 not found.
SIP: (209) Attribute mid, level 2 instance 1 not found.
Mar 18 08:14:19.189: //209/9C38ACFE822C/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entry
Mar 18 08:14:19.189: //209/9C38ACFE822C/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:failed to update call entry
Mar 18 08:14:19.189: //209/9C38ACFE822C/SIP/Error/sipSPI_ipip_update_call_entry:failed to update call entry
Mar 18 08:14:19.189: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 209
Mar 18 08:14:19.189: //209/9C38ACFE822C/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
Mar 18 08:14:19.197: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_call_service_msg: ccb NULL, unable to update the callinfo ui parameters
Mar 18 08:14:19.197: //209/9C38ACFE822C/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
Mar 18 08:14:20.641: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed
Mar 18 08:14:20.641: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed
Mar 18 08:14:20.725: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_call_service_msg: ccb NULL, unable to update the callinfo ui parameters
Mar 18 08:14:20.729: %DSMP-3-DSPALARM: Alarm on DSP : status=0x0 message=0x0 text=N
Mar 18 08:14:20.729: //209/9C38ACFE822C/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode SIP: (209) Group (a= group line) attribute, level 65535 instance 1 not found.
Mar 18 08:14:20.733: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIUcccause_to_sipcause: Out of Range PSTN Cause Code from CCAPI: 172
Mar 18 08:14:20.765: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length
Mar 18 08:14:20.929: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length
Mar 18 08:14:30.653: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashed
Mar 18 08:14:30.653: //-1/xxxxxxxxxxxx/SIP/Error/HandleUdpIPv4SocketReads: SIP Message incomplete, trashedno debug ccsip error
SIP Call error tracing is disabled