I would use TCP transport with Cisco routers. they like it better unless you tickle them.
You read my mind, I was just trying TCP but had no luck at all … I changed the transport on the PJSIP trunk (the one i just made, I’m still using that atm) and swapped the transport on the outgoing dial-peer but it complained that it couldn’t reach Asterisk at all.
I tried splitting up the dial-peers so the incoming call wasn’t matching the dial-peer that had tcp set as the transport type. It should’ve come in as UDP, then the bridge to Asterisk should’ve been TCP. But yeah, got disconnect cause 38 … didn’t even looked like it tried to send anything to Asterisk though …
Edit: the Cisco disconnects the call when it matches the outgoing dial-peer, before it even tried to send an INVITE… so yeah…
Gotta love cisco’s , can the FreeSwitch talk to the Cisco yet ? (start at both ends, to divide and conquer in half the time) Because Cause 38 means “the network is out of order” So back to your cisco RTFM’s 
Yeah the error 38 was the cause in relation to the TCP PJSIP bridge to Asterisk. FS is talking to the Cisco just fine with UDP and sofia SIP.
Yeah… Cisco VOIP is just … greeaat
/sarcasm
I would just ditch the Cisco border if it weren’t for the fact that I’ll need to terminate our PRI on it, and trunk that over to Asterisk.
Edit: I’m not going to chase problems with TCP. It should work with UDP, and is working with UDP to an Asterisk 11 server which I am migrating away from.
And not try chan_sip as a check?, IWFFM . . .
@dicko What do you know about DSP’s in Cisco?
Perhaps this is the issue, though, it’s not a very busy circuit at the moment - it is taking calls to some of our local-rate 1300 numbers, that are passed over to an older server.
When i do a show dsp all or show dspfarm all both times it comes back and says;
voice-gw3.staff#show dspfarm all
DSPFARM Configuration Information:
Admin State: DOWN, Oper Status: DOWN - Cause code: ADMIN_STATE_DOWN
Transcoding Sessions: 0(Avail: 0), Conferencing Sessions: 0 (Avail: 0)
Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 600
Connection check interval 600 Codec G729 VAD: ENABLED
voice-gw3.staff#show dsp all
DSPFARM Configuration Information:
Admin State: DOWN, Oper Status: DOWN - Cause code: ADMIN_STATE_DOWN
Transcoding Sessions: 0(Avail: 0), Conferencing Sessions: 0 (Avail: 0)
Trans sessions for mixed-mode conf: 0 (Avail: 0), RTP Timeout: 600
Connection check interval 600 Codec G729 VAD: ENABLED
To me, that looks like there are no DSP’s available to transcode. I’m trying not to transcode, all calls should come up either with ULAW and ALAW but I believe this still requires DSP resources. Doesn’t explain why it’s working with the old server though, if DSP’s were the problem it should’ve never worked imo…
I was using chan_sip originally with the same problem.
Can you move chan_sip over to TCP?
Yes you can use TCP over chan_sip, just enable and prefer it . . .
It’s a shame that @SkykingOH is no longer with us, he would sort you out in a second.
So, at the risk of copping an RTFM from you …
I added this to my chan_sip trunk config;
tcpenable=yes
tcpbind=0.0.0.0
But I don’t think it’s right. I don’t want to do anything that may break other chan_sip trunks that are currently working with UDP (I have a trunk to a Cisco Unified CallManager, using chan_sip on 5061 with no problems).
Anyway, the Cisco really doesn’t like TCP - at some point before it INVITE’s the second leg it gives up with the error 38 before, so it already knows that it can’t get to the server with TCP. Pretty sure this isn’t a firewall iptables problem.
yes, you enabled it (rather loosely with 0.0.0.0) just use that as a transport for the “extension” , by default it will use UDP.
You will always get a 38 until the “network” is working.
So, can I have this;
ITSP <–udp–> Cisco <–tcp chan_sip trunk–> Asterisk <–udp–> handset
I’m not registering the Cisco border to Asterisk as an Extension - it’s just setup as a trunk with no register string.
Today we are going to try an IOS upgrade on the router followed by restarting the Asterisk service. I’ll post back my results.
This didn’t work…