sip.conf code is as follows: [101_5] type=peer secret=******* host=dynamic defaultip=192.168.0.93 username=101_5 qualify=yes context=garsongold_out disallow=all allow=ulaw mailbox=101@garsongold [101_5] type=user secret=******* host=dynamic ; This phone belongs to callgroup 1 callgroup=1 ; This phone can pick up anyone in callgroup 1 ;pickupgroup=1 defaultip=192.168.0.93 username=101_5 qualify=yes context=garsongold_out disallow=all allow=ulaw mailbox=101@garsongold callerid="Reception" <101> Dialplan code is as follows: [garsongold_out] exten => _X.,1,Noop(Dialed extension ${EXTEN}.) exten => _X.,n,Goto(garsongold_out_tmp,${EXTEN},1) exten => _*X.,1,Noop(Dialed extension ${EXTEN}.) exten => _*X.,n,Goto(garsongold_out_tmp,${EXTEN},1) [garsongold_out_tmp] include => garsongold_extensions include => conferences include => garsongold_local_out include => ld-out include => ild-out include => check-voicemail SIP Debug follows: 1061 <--- SIP read from 192.168.0.93:5060 ---> 1062 INVITE sip:6047283777@192.168.0.25 SIP/2.0^M 1063 From: ;tag=bbeb232cdcc4048^M 1064 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1065 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bKd3e2f2b69^M 1066 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1067 CSeq: 708817711 INVITE^M 1068 Max-Forwards: 70^M 1069 Content-Length: 237^M 1070 Supported: timer^M 1071 Allow-Events: talk,hold,conference^M 1072 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M 1073 Content-Type: application/sdp^M 1074 Supported: replaces^M 1075 Contact: ^M 1076 User-Agent: Aastra 57i/2.0.2.1040 Brcm-Callctrl/v1.7.2.2 M5T SIP-UA SAFE/v3.6.7.11^M 1077 ^M 1078 v=0^M 1079 o=MxSIP 0 1122476827 IN IP4 192.168.0.93^M 1080 s=SIP Call^M 1081 c=IN IP4 0.0.0.0^M 1082 t=0 0^M 1083 m=audio 3000 RTP/AVP 0 101^M 1084 a=rtpmap:0 PCMU/8000^M 1085 a=rtpmap:101 telephone-event/8000^M 1086 a=fmtp:101 0-15^M 1087 a=ptime:20^M 1088 a=sendonly^M 1089 a=silenceSupp:off - - - -^M 1090 1091 <-------------> 1092 [Jan 30 09:31:44] VERBOSE[18487] logger.c: --- (15 headers 12 lines) --- 1093 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Sending to 192.168.0.93 : 5060 (no NAT) 1094 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Found RTP audio format 0 1095 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Found RTP audio format 101 1096 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Peer audio RTP is at port 0.0.0.0:3000 1097 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Found audio description format PCMU for ID 0 1098 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Found audio description format telephone-event for ID 101 1099 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) 1100 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (tele phone-event) 1101 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Peer audio RTP is at port 0.0.0.0:3000 1102 [Jan 30 09:31:44] VERBOSE[18487] logger.c:� 1103 <--- Transmitting (no NAT) to 192.168.0.93:5060 ---> 1104 SIP/2.0 100 Trying^M 1105 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bKd3e2f2b69;received=192.168.0.93^M 1106 From: ;tag=bbeb232cdcc4048^M 1107 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1108 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1109 CSeq: 708817711 INVITE^M 1110 User-Agent: Asterisk PBX^M 1111 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M 1112 Supported: replaces^M 1113 Contact: ^M 1114 Content-Length: 0^M 1115 ^M 1116 1117 <------------> 1118 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Audio is at 192.168.0.25 port 16160 1119 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Adding codec 0x4 (ulaw) to SDP 1120 [Jan 30 09:31:44] VERBOSE[18487] logger.c: Adding non-codec 0x1 (telephone-event) to SDP 1121 [Jan 30 09:31:44] VERBOSE[18487] logger.c:� 1122 <--- Transmitting (no NAT) to 192.168.0.93:5060 ---> 1123 SIP/2.0 200 OK^M 1124 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bKd3e2f2b69;received=192.168.0.93^M 1125 From: ;tag=bbeb232cdcc4048^M 1126 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1127 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1128 CSeq: 708817711 INVITE^M 1129 User-Agent: Asterisk PBX^M 1130 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M 1131 Supported: replaces^M 1132 Contact: ^M 1133 Content-Type: application/sdp^M 1134 Content-Length: 240^M 1135 ^M 1136 v=0^M 1137 o=root 18478 18479 IN IP4 192.168.0.25^M 1138 s=session^M 1139 c=IN IP4 192.168.0.25^M 1140 t=0 0^M 1141 m=audio 16160 RTP/AVP 0 101^M 1142 a=rtpmap:0 PCMU/8000^M 1143 a=rtpmap:101 telephone-event/8000^M 1144 a=fmtp:101 0-16^M 1145 a=silenceSupp:off - - - -^M 1146 a=ptime:20^M 1147 a=recvonly^M 1148 1149 <------------> 1150 [Jan 30 09:31:44] VERBOSE[18882] logger.c: -- Started music on hold, class 'default', on Zap/7-1 1151 [Jan 30 09:31:45] VERBOSE[18487] logger.c:� 1152 <--- SIP read from 192.168.0.93:5060 ---> 1153 ACK sip:6047283777@192.168.0.25 SIP/2.0^M 1154 From: ;tag=bbeb232cdcc4048^M 1155 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1156 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bKe82178182^M 1157 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1158 CSeq: 708817711 ACK^M 1159 Max-Forwards: 70^M 1160 Content-Length: 0^M 1161 User-Agent: Aastra 57i/2.0.2.1040 Brcm-Callctrl/v1.7.2.2 M5T SIP-UA SAFE/v3.6.7.11^M 1162 ^M 1163 1164 <-------------> 1165 [Jan 30 09:31:45] VERBOSE[18487] logger.c: --- (9 headers 0 lines) --- 1166 [Jan 30 09:31:48] VERBOSE[18487] logger.c:� 1167 <--- SIP read from 192.168.0.93:5060 ---> 1168 REFER sip:6047283777@192.168.0.25 SIP/2.0^M 1169 From: ;tag=bbeb232cdcc4048^M 1170 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1171 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bK58151768f^M 1172 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1173 CSeq: 708817712 REFER^M 1174 Max-Forwards: 70^M 1175 Content-Length: 0^M 1176 Supported: timer^M 1177 Allow-Events: talk,hold,conference^M 1178 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M 1179 Contact: ^M 1180 Refer-To: ^M 1181 Referred-By: ^M 1182 Supported: replaces^M 1183 User-Agent: Aastra 57i/2.0.2.1040 Brcm-Callctrl/v1.7.2.2 M5T SIP-UA SAFE/v3.6.7.11^M 1184 ^M 1185 1186 <-------------> 1187 [Jan 30 09:31:48] VERBOSE[18487] logger.c: --- (16 headers 0 lines) --- 1188 [Jan 30 09:31:48] VERBOSE[18487] logger.c: Call 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25 got a SIP call transfer from caller: (REFER)! 1189 [Jan 30 09:31:48] VERBOSE[18487] logger.c: Failed SIP Transfer to non-existing extension 192.168.0.25:5060 in context garsongold_out 1190 n[Jan 30 09:31:48] VERBOSE[18487] logger.c:� 1191 <--- Transmitting (no NAT) to 192.168.0.93:5060 ---> 1192 SIP/2.0 202 Accepted^M 1193 Via: SIP/2.0/UDP 192.168.0.93;branch=z9hG4bK58151768f;received=192.168.0.93^M 1194 From: ;tag=bbeb232cdcc4048^M 1195 To: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1196 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1197 CSeq: 708817712 REFER^M 1198 User-Agent: Asterisk PBX^M 1199 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M 1200 Supported: replaces^M 1201 Contact: ^M 1202 Content-Length: 0^M 1203 ^M 1204 1205 <------------> 1206 [Jan 30 09:31:48] VERBOSE[18487] logger.c: set_destination: Parsing for address/port to send to 1207 [Jan 30 09:31:48] VERBOSE[18487] logger.c: set_destination: set destination to 192.168.0.93, port 5060 1208 [Jan 30 09:31:48] VERBOSE[18487] logger.c: Reliably Transmitting (no NAT) to 192.168.0.93:5060: 1209 NOTIFY sip:101_5@192.168.0.93;transport=udp SIP/2.0^M 1210 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK12a971e0;rport^M 1211 From: "BRIT. COLUMBIA " ;tag=as448dcd4a^M 1212 To: ;tag=bbeb232cdcc4048^M 1213 Contact: ^M 1214 Call-ID: 4ee474343a6b4ab57166f58b7c5919be@192.168.0.25^M 1215 CSeq: 103 NOTIFY^M 1216 User-Agent: Asterisk PBX^M 1217 Max-Forwards: 70^M 1218 Event: refer;id=708817712^M 1219 Subscription-state: terminated;reason=noresource^M 1220 Content-Type: message/sipfrag;version=2.0^M 1221 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M 1222 Supported: replaces^M 1223 Content-Length: 23^M