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Is it possible you are exceeding the number of available channels at the time you are getting these errors? ... or, did you try to dial those numbers from your cellphone to see if they are actually valid phone numbers?
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Is it possible you are exceeding the number of available channels at the time you are getting these errors? ... or, did you try to dial those numbers from your cellphone to see if they are actually valid phone numbers?

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Also, check to see which way the Telco is hunting and which way you are hunting to minimize glare. (both sides trying to seize the same channel at the same time)<br />
It doesn't matter which side hunts top down and which one hunts bottom up, as long as they are opposite.
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Also, check to see which way the Telco is hunting and which way you are hunting to minimize glare. (both sides trying to seize the same channel at the same time)
It doesn't matter which side hunts top down and which one hunts bottom up, as long as they are opposite.

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sing SIP RTP CoS mark 5<br />
&nbsp; &nbsp; -- Executing [09599259123@numberplan2:1<wbr />] Dial(&quot;SIP/220-00000015&quot;, &quot;DAHDI/G1/9599259123,60&quot;) in new stack<br />
[Jun 13 17:19:19] WARNING[3910]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)<br />
&nbsp; == Everyone is busy/congested at this time (1:0/1/0)<br />
&nbsp; &nbsp; -- Executing [09599259123@numberplan2:2<wbr />] Hangup(&quot;SIP/220-00000015&quot;,<wbr /> &quot;&quot;) in new stack
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sing SIP RTP CoS mark 5
    -- Executing [09599259123@numberplan2:1] Dial("SIP/220-00000015", "DAHDI/G1/9599259123,60") in new stack
[Jun 13 17:19:19] WARNING[3910]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [09599259123@numberplan2:2] Hangup("SIP/220-00000015", "") in new stack

pri not working on pbx some extensions are working some are not

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

IP Telephony

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).