<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt">Hi,<br>I am hoping someone can help me with a problem I am having.<br>I am trying to setup a connection from an Elastix 2 server to a Siemens isdx PBX. The setup is as follows<br><br>Elastix 2<br> *sip trunk*<br>Cisco 2621XM router with 2 E1 voice interfaces<br> *QSIG*<br>Westell IQ2000 protocol convertor<br> *DPNSS*<br>Siemens ISDX<br><br>So the Elastix box has a SIP trunk to the cisco router which then talks QSIG to the Westell which converts it to DPNSS to talk to the ISDX.<br>I have managed to make a call from a phone on Elastix to a phone on the ISDX but it drops after about 3 seconds, every time. Would anyone have any idea why this is? <br>Here is the setup I have on Elastix and the Cisco router<br><br>Elastix SIP trunk PEER
details<br>type=friend<br>qualify=no<br>nat=no<br>insecure=very<br>host=10.132.41.13<br>dtmfmode=rfc2833<br>context=from-internal<br>canreinvite=yes<br>disallow=all<br>allow=ulaw<br><br>relevant Cisco config<br>version 12.4<br>service timestamps debug datetime msec<br>service timestamps log datetime msec<br>service password-encryption<br>!<br>boot-start-marker<br>boot-end-marker<br>!<br>no network-clock-participate slot 1<br>no network-clock-participate wic 0<br>ip cef<br>!<br>!<br>no ip domain lookup<br>isdn switch-type primary-qsig<br>voice-card 1<br>!<br>!<br>voice rtp send-recv<br>!<br>voice service voip<br> allow-connections h323 to h323<br> allow-connections h323 to sip<br> allow-connections sip to h323<br> allow-connections sip to sip<br> fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco<br> sip<br> bind control source-interface FastEthernet0/0<br> bind media source-interface
FastEthernet0/0<br>!<br>!<br>voice class codec 1<br> codec preference 1 g711ulaw<br> codec preference 2 g729r8 bytes 40<br>!<br>!<br>controller E1 1/0<br>!<br>controller E1 1/1<br> pri-group timeslots 1-31<br>!<br>!<br>class-map match-all FAX<br> description Match T.38 Fax<br> match access-group name FAX<br>!<br>interface FastEthernet0/0<br> ip address 10.132.41.13 255.255.255.0<br> no ip mroute-cache<br> speed 100<br> full-duplex<br>!<br>interface FastEthernet0/1<br> no ip address<br> duplex auto<br> speed auto<br>!<br>interface Serial1/1:15<br> no ip address<br> encapsulation hdlc<br> no logging event link-status<br> isdn switch-type primary-qsig<br> isdn overlap-receiving<br> isdn incoming-voice voice<br> no cdp enable<br>!<br>ip forward-protocol nd<br>ip route 0.0.0.0 0.0.0.0 10.132.41.1<br>!<br>ip http server<br>!<br>control-plane<br>!<br>voice-port
1/1:15<br>!<br>!<br>dial-peer voice 786 voip<br> huntstop<br> destination-pattern 56...<br> progress_ind setup enable 3<br> voice-class codec 1<br> session protocol sipv2<br> session target sip-server<br> dtmf-relay rtp-nte<br> no vad<br>!<br>dial-peer voice 2 pots<br> description QSIG Link to PABX via Westell<br> destination-pattern 2....<br> progress_ind setup enable 3<br> progress_ind alert enable 8<br> direct-inward-dial<br> port 1/1:15<br> forward-digits all<br>!<br>dial-peer voice 3 pots<br> description Featurenet 7xx routing<br> huntstop<br> destination-pattern 7T<br> progress_ind setup enable 3<br> progress_ind alert enable 8<br> incoming called-number 786....<br> no digit-strip<br> direct-inward-dial<br> forward-digits all<br>!<br>gateway<br>!<br>sip-ua<br> registrar ipv4:10.133.40.50 expires 3600<br> sip-server
ipv4:10.133.40.50<br><br>Any help would be greatly appreciated<br>Thanks,<br>DT<br><div><br></div>
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