[asterisk-ss7] Circuit/channel congestion

Markus A. Wipfler markus at infocom.co.ug
Wed Dec 3 00:09:56 CST 2008

Hi Group,

some advise needed. I am doing International call termination using  
Asterisk/Sangoma/Zaptel. My setup is as follows:

voippartners (internet)
asteriskbox(sip only, high end server, cdr database, monitoring  
scripts, call routing based on destination)-------sip----- 
terminationbox1 (asterisk/libss7/sangoma)----ss7-----telecom1
terminationboxN (Asterisk/libss7/sangoma)

I had many issues to get above setup up and running (from  
misconfigured zaptel  to miss mapped CIC on the telecom switch), but  
now i have reached a point where I do about 40K call minutes per day  
on each of the
termination boxes and this project is finally starting to pay off.  
However i have noticed that of late I am getting the below warnings a  

WARNING[23136] app_dial.c: Unable to create channel of type  
'Zap' (cause 34 - Circuit/channel congestion)

Am getting this warnings even when I have enough channel capacity on  
my E1s. During this time SIP channels still work and i dont see any  
complaints from the kernel. The call termination box gets the call in  
via sip but asterisk is not able to push it to the zaptel driver it  
seems. Yet when this starts happening it doesn't seem to affect all  
new calls. Any help is appreciated.


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