Marcel V Posted December 4, 2008 Report Posted December 4, 2008 I have a sucessfull outboud call with PBXNSIP 3, but the inboud call does not work The error is: [8] 2008/12/04 18:06:42: Could not find a trunk (2 trunks) [8] 2008/12/04 18:06:42: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header [9] 2008/12/04 18:06:42: Resolve 69885: udp 217.71.123.3 5060 [6] 2008/12/04 18:06:42: Sending RTP for 1792407011@217.71.123.10#8b444b73ec to 217.71.123.11:14270 [5] 2008/12/04 18:06:42: Received incoming call without trunk information and user has not been found The trunk is setup for in and outboud. What do i have to change in the truck setup to support inbound calls ? Quote
jlumby Posted December 4, 2008 Report Posted December 4, 2008 I have a sucessfull outboud call with PBXNSIP 3, but the inboud call does not work The error is: [8] 2008/12/04 18:06:42: Could not find a trunk (2 trunks) [8] 2008/12/04 18:06:42: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header [9] 2008/12/04 18:06:42: Resolve 69885: udp 217.71.123.3 5060 [6] 2008/12/04 18:06:42: Sending RTP for 1792407011@217.71.123.10#8b444b73ec to 217.71.123.11:14270 [5] 2008/12/04 18:06:42: Received incoming call without trunk information and user has not been found The trunk is setup for in and outboud. What do i have to change in the truck setup to support inbound calls ? What do you have "Send call to extension:" set to? Quote
Marcel V Posted December 4, 2008 Author Report Posted December 4, 2008 What do you have "Send call to extension:" set to? To an extension "40" Quote
shopcomputer Posted December 4, 2008 Report Posted December 4, 2008 To an extension "40" How many trunks do you have, it looks like you have 2 trunks, is the other also set to send to 40? Quote
Marcel V Posted December 12, 2008 Author Report Posted December 12, 2008 How many trunks do you have, it looks like you have 2 trunks, is the other also set to send to 40? Your right, but we have only 1 trunk. Quote
shopcomputer Posted December 12, 2008 Report Posted December 12, 2008 Your right, but we have only 1 trunk. Try adding your phone number as as alias on extension 40. Quote
Marcel V Posted December 12, 2008 Author Report Posted December 12, 2008 Try adding your phone number as as alias on extension 40. Hi same case [8] 2008/12/12 17:22:39: Could not find a trunk (1 trunks) [8] 2008/12/12 17:22:39: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header [9] 2008/12/12 17:22:39: Resolve 23: udp 217.71.123.3 5060 [6] 2008/12/12 17:22:39: Sending RTP for 151170848@217.71.123.10#fab816e05e to 217.71.123.12:12950 [5] 2008/12/12 17:22:39: Received incoming call without trunk information and user has not been found I found the trunk problem, was 2 now 1 as you can see. But i do not understand why it does not sent the incoming calls to extension 40 Quote
shopcomputer Posted December 12, 2008 Report Posted December 12, 2008 Hi same case [8] 2008/12/12 17:22:39: Could not find a trunk (1 trunks) [8] 2008/12/12 17:22:39: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header [9] 2008/12/12 17:22:39: Resolve 23: udp 217.71.123.3 5060 [6] 2008/12/12 17:22:39: Sending RTP for 151170848@217.71.123.10#fab816e05e to 217.71.123.12:12950 [5] 2008/12/12 17:22:39: Received incoming call without trunk information and user has not been found I found the trunk problem, was 2 now 1 as you can see. But i do not understand why it does not sent the incoming calls to extension 40 Something must be wrong in there, you can call me I am sure I can resolve it, with a quick phone call. 888.IPPBX.US Quote
Marcel V Posted December 16, 2008 Author Report Posted December 16, 2008 Something must be wrong in there, you can call me I am sure I can resolve it, with a quick phone call. 888.IPPBX.US HI, Thanks for the offer, I changed SIP provider and now it works, BUT the outbounds now does not work. Here is my log: [9] 2008/12/16 15:37:19: UDP: Opening socket on port 49430 [9] 2008/12/16 15:37:19: UDP: Opening socket on port 49431 [8] 2008/12/16 15:37:19: Could not find a trunk (2 trunks) [8] 2008/12/16 15:37:19: Using outbound proxy sip:195.73.157.235:31268;transport=udp because UDP packet source did not match the via header [9] 2008/12/16 15:37:19: Resolve 3369: udp 195.73.157.235 31268 [9] 2008/12/16 15:37:19: Resolve 3370: udp 195.73.157.235 31268 [8] 2008/12/16 15:37:20: Tagging request with existing tag [6] 2008/12/16 15:37:20: Sending RTP for NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508 to 192.168.3.221:33392 [9] 2008/12/16 15:37:20: Resolve 3371: udp 195.73.157.235 31268 [9] 2008/12/16 15:37:20: Dialplan: Evaluating !^0([0-9]*)@.*!sip:0031\1@\r;user=phone!i against 0345549745@195.73.157.237 [5] 2008/12/16 15:37:20: Dialplan Standaard: Match 0345549745@195.73.157.237 to <sip:0031345549745@sip.goandcall.com;user=phone> on trunk Xeloq [8] 2008/12/16 15:37:20: Play audio_moh/noise.wav [9] 2008/12/16 15:37:20: UDP: Opening socket on port 60242 [9] 2008/12/16 15:37:20: UDP: Opening socket on port 60243 [9] 2008/12/16 15:37:20: Resolve 3372: url sip:sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3372: naptr sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3372: srv tls _sips._tcp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3372: srv tcp _sip._tcp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3372: srv udp _sip._udp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3372: a udp sip.goandcall.com 5060 [9] 2008/12/16 15:37:20: Resolve 3372: udp 81.26.212.150 5060 [9] 2008/12/16 15:37:20: Resolve 3373: udp 195.73.157.235 31268 [8] 2008/12/16 15:37:20: Answer challenge with username 725319 [9] 2008/12/16 15:37:20: Resolve 3374: udp 81.26.212.150 5060 udp:1 [9] 2008/12/16 15:37:20: Resolve 3375: udp 81.26.212.150 5060 udp:1 [9] 2008/12/16 15:37:20: Message repetition, packet dropped [6] 2008/12/16 15:37:20: Sending RTP for NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508 to 195.73.157.235:33392 [7] 2008/12/16 15:37:20: Call 6b86f62f@pbx#3710: Clear last INVITE [9] 2008/12/16 15:37:20: Resolve 3376: url sip:sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3376: naptr sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3376: srv tls _sips._tcp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3376: srv tcp _sip._tcp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3376: srv udp _sip._udp.sip.goandcall.com [9] 2008/12/16 15:37:20: Resolve 3376: a udp sip.goandcall.com 5060 [9] 2008/12/16 15:37:20: Resolve 3376: udp 81.26.212.150 5060 [5] 2008/12/16 15:37:20: INVITE Response: Terminate 6b86f62f@pbx [7] 2008/12/16 15:37:20: Other Ports: 1 [7] 2008/12/16 15:37:20: Call Port: NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508 [9] 2008/12/16 15:37:20: Resolve 3377: udp 195.73.157.235 31268 [9] 2008/12/16 15:37:23: Resolve 3378: udp 195.73.157.235 61639 [9] 2008/12/16 15:37:23: Message repetition, packet dropped Appreciate the feedback, Quote
Vodia PBX Posted December 16, 2008 Report Posted December 16, 2008 Can you turn SIP packet logging on? You should see a packet with the header "INVITE sip:xxx SIP/2.0" (see http://wiki.pbxnsip.com/index.php/Log_Setup on how to do that). Probably the SIP provider rejects the call. Quote
Marcel V Posted December 17, 2008 Author Report Posted December 17, 2008 Can you turn SIP packet logging on? You should see a packet with the header "INVITE sip:xxx SIP/2.0" (see http://wiki.pbxnsip.com/index.php/Log_Setup on how to do that). Probably the SIP provider rejects the call. I have xlite connected to PBXNSIP to call and changed this xlite client to connect directly to the sip provider instead of the PBXNSIP and this works !! ?? Quote
pbx support Posted December 17, 2008 Report Posted December 17, 2008 I have xlite connected to PBXNSIP to call and changed this xlite client to connect directly to the sip provider instead of the PBXNSIP and this works !! ?? Still SIP log would tell us what is wrong. SIP messaging will be different when x-lite is directly connected to the provider vs connected via the PBX Quote
Marcel V Posted December 19, 2008 Author Report Posted December 19, 2008 Still SIP log would tell us what is wrong. SIP messaging will be different when x-lite is directly connected to the provider vs connected via the PBX I have attached the file with the log, i have tried to dial out twice. I have a dialplan attached as picture and my trunks. MV2008_12_191.txt Quote
Vodia PBX Posted December 19, 2008 Report Posted December 19, 2008 I have attached the file with the log, i have tried to dial out twice. I have a dialplan attached as picture and my trunks. The "GNC SS5 Proxy" replies with 603 Declined. I assume that is because the authentication information is in the P-Asserted-Identity header. Try using "Remote Party/Privacy Indication" in the trunk "No indication" Quote
Marcel V Posted December 22, 2008 Author Report Posted December 22, 2008 The "GNC SS5 Proxy" replies with 603 Declined. I assume that is because the authentication information is in the P-Asserted-Identity header. Try using "Remote Party/Privacy Indication" in the trunk "No indication" Thanks, that did the trick Quote
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