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No path to translate from SIP/logosoft-00000003(256) to SIP/

http://forum.eflo.net/VICIDIALforum/viewtopic.php?f=7&t=26702

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No path to translate from SIP/logosoft-00000003(256) to SIP/

PostPosted: Sat Jan 12, 2013 8:17 am
by scenarist
I have the problem with make calls via Voip service provider based on IP authentification.
I have successfully registered it but I still can not make outgoing calls.

This is my CLI: (PLEASE pay attention on WARNING! What it means ? , some advice ?)
Code: Select all
[Jan 12 07:26:36]     -- Executing [9062723254@default:1] AGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 12 07:26:36]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 12 07:26:36]     -- Executing [9062723254@default:2] Dial("SIP/8001-00000002", "SIP/062723254@logosoft||tTo") in new stack
[Jan 12 07:26:36]     -- Called 062723254@logosoft
[Jan 12 07:26:36] WARNING[23806]: channel.c:3908 ast_channel_make_compatible: No path to translate from SIP/logosoft-00000003(256) to SIP/8001-00000002(4)
[Jan 12 07:26:36]   == Spawn extension (default, 9062723254, 2) exited non-zero on 'SIP/8001-00000002'
[Jan 12 07:26:36]     -- Executing [h@default:1] DeadAGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION----------") in new stack
[Jan 12 07:26:36]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION---------- completed, returning 0


This is result of command sip show peers

Code: Select all
go*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
8001/8001                  217.75.206.98    D   N      22012    OK (113 ms)
logosoft                   217.75.205.49        N      5060     OK (7 ms)


This is my configuration of sip carrier.
Image
and
this is sip set debug ip

Code: Select all
[Jan 12 07:34:08] --- (14 headers 18 lines) ---
[Jan 12 07:34:08] Really destroying SIP dialog '4ca3786e0329a78c503c944d75d2c3b0@217.75.206.97' Method: OPTIONS
[Jan 12 07:34:43]     -- Executing [9062723254@default:1] AGI("SIP/8001-00000006", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 12 07:34:43]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 12 07:34:43]     -- Executing [9062723254@default:2] Dial("SIP/8001-00000006", "SIP/062723254@logosoft||tTo") in new stack
[Jan 12 07:34:43] Audio is at 217.75.206.97 port 15780
[Jan 12 07:34:43] Adding codec 0x4 (ulaw) to SDP
[Jan 12 07:34:43] Adding codec 0x2 (gsm) to SDP
[Jan 12 07:34:43] Adding non-codec 0x1 (telephone-event) to SDP
[Jan 12 07:34:43] Reliably Transmitting (NAT) to 217.75.205.49:5060:
INVITE sip:062723254@217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>
Contact: <sip:0000000000@217.75.206.97>
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:0000000000@217.75.206.97>;privacy=off;screen=no
Date: Sat, 12 Jan 2013 12:34:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 2441 2441 IN IP4 217.75.206.97
s=session
c=IN IP4 217.75.206.97
t=0 0
m=audio 15780 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Jan 12 07:34:43]     -- Called 062723254@logosoft
[Jan 12 07:34:43] WARNING[28119]: channel.c:3908 ast_channel_make_compatible: No path to translate from SIP/logosoft-00000007(256) to SIP/8001-00000006(4)
[Jan 12 07:34:43] Scheduling destruction of SIP dialog '785a880422f10658673123852317ad1b@217.75.206.97' in 6400 ms (Method: INVITE)
[Jan 12 07:34:43]
<--- SIP read from 217.75.205.49:55203 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>
Date: Sat, 12 Jan 2013 12:34:42 GMT
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


<------------->
[Jan 12 07:34:43] --- (10 headers 0 lines) ---
[Jan 12 07:34:43] Reliably Transmitting (NAT) to 217.75.205.49:55203:
CANCEL sip:062723254@217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:0000000000@217.75.206.97>;privacy=off;screen=no
Content-Length: 0


---
[Jan 12 07:34:43] Scheduling destruction of SIP dialog '785a880422f10658673123852317ad1b@217.75.206.97' in 6400 ms (Method: INVITE)
[Jan 12 07:34:43]
<--- SIP read from 217.75.205.49:55203 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>
Date: Sat, 12 Jan 2013 12:34:42 GMT
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 CANCEL
Content-Length: 0


<------------->
[Jan 12 07:34:43] --- (8 headers 0 lines) ---
[Jan 12 07:34:43]
<--- SIP read from 217.75.205.49:55203 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>;tag=D8B4A1F4-5FE
Date: Sat, 12 Jan 2013 12:34:42 GMT
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0


<------------->
[Jan 12 07:34:43] --- (11 headers 0 lines) ---
[Jan 12 07:34:43] Transmitting (NAT) to 217.75.205.49:55203:
ACK sip:062723254@217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK0a159022;rport
From: "8001" <sip:0000000000@217.75.206.97>;tag=as1928fc53
To: <sip:062723254@217.75.205.49;cpd=on>;tag=D8B4A1F4-5FE
Contact: <sip:0000000000@217.75.206.97>
Call-ID: 785a880422f10658673123852317ad1b@217.75.206.97
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "8001" <sip:0000000000@217.75.206.97>;privacy=off;screen=no
Content-Length: 0


---
[Jan 12 07:34:43]   == Spawn extension (default, 9062723254, 2) exited non-zero on 'SIP/8001-00000006'
[Jan 12 07:34:43]     -- Executing [h@default:1] DeadAGI("SIP/8001-00000006", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION----------") in new stack
[Jan 12 07:34:43] Really destroying SIP dialog '785a880422f10658673123852317ad1b@217.75.206.97' Method: INVITE
[Jan 12 07:34:43]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CONGESTION---------- completed, returning 0
[Jan 12 07:35:03]   == Parsing '/etc/asterisk/manager.conf': [Jan 12 07:35:03] Found
[Jan 12 07:35:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 12 07:35:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 12 07:35:03]   == Parsing '/etc/asterisk/manager.conf': [Jan 12 07:35:03] Found
[Jan 12 07:35:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 12 07:35:05]  Reloading SIP
[Jan 12 07:35:05]   == Parsing '/etc/asterisk/sip.conf': [Jan 12 07:35:05] Found
[Jan 12 07:35:05]   == Parsing '/etc/asterisk/sip-vicidial.conf': [Jan 12 07:35:05] Found
[Jan 12 07:35:05]   == Parsing '/etc/asterisk/sip_notify.conf': [Jan 12 07:35:05] Found
[Jan 12 07:35:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 12 07:35:07] Reliably Transmitting (NAT) to 217.75.205.49:5060:
OPTIONS sip:217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK73a8663b;rport
From: "asterisk" <sip:asterisk@217.75.206.97>;tag=as1f10f0b3
To: <sip:217.75.205.49;cpd=on>
Contact: <sip:asterisk@217.75.206.97>
Call-ID: 7f3cb8b4673843c664bf3044541b5f8c@217.75.206.97
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 12 Jan 2013 12:35:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Jan 12 07:35:07]
<--- SIP read from 217.75.205.49:55203 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK73a8663b;rport
From: "asterisk" <sip:asterisk@217.75.206.97>;tag=as1f10f0b3
To: <sip:217.75.205.49;cpd=on>;tag=D8B5002C-182
Date: Sat, 12 Jan 2013 12:35:06 GMT
Call-ID: 7f3cb8b4673843c664bf3044541b5f8c@217.75.206.97
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 452

v=0
o=CiscoSystemsSIP-GW-UserAgent 8329 7454 IN IP4 217.75.205.49
s=SIP Call
c=IN IP4 217.75.205.49
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 217.75.205.49
m=image 0 udptl t38
c=IN IP4 217.75.205.49
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[Jan 12 07:35:07] --- (14 headers 18 lines) ---
[Jan 12 07:35:07] Really destroying SIP dialog '7f3cb8b4673843c664bf3044541b5f8c@217.75.206.97' Method: OPTIONS
[Jan 12 07:35:08]   == Parsing '/etc/asterisk/manager.conf': [Jan 12 07:35:08] Found
[Jan 12 07:35:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 12 07:35:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 12 07:35:28]   == Refreshing DNS lookups.
[Jan 12 07:36:03]   == Parsing '/etc/asterisk/manager.conf': [Jan 12 07:36:03] Found
[Jan 12 07:36:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 12 07:36:03]   == Parsing '/etc/asterisk/manager.conf': [Jan 12 07:36:03] Found
[Jan 12 07:36:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 12 07:36:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 12 07:36:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 12 07:36:07] Reliably Transmitting (NAT) to 217.75.205.49:5060:
OPTIONS sip:217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK1d5190ff;rport
From: "asterisk" <sip:asterisk@217.75.206.97>;tag=as59136e49
To: <sip:217.75.205.49;cpd=on>
Contact: <sip:asterisk@217.75.206.97>
Call-ID: 273684113831d61a4520dcff39b44004@217.75.206.97
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 12 Jan 2013 12:36:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Jan 12 07:36:07]
<--- SIP read from 217.75.205.49:55203 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.75.206.97:5060;branch=z9hG4bK1d5190ff;rport
From: "asterisk" <sip:asterisk@217.75.206.97>;tag=as59136e49
To: <sip:217.75.205.49;cpd=on>;tag=D8B5EA94-1DD8
Date: Sat, 12 Jan 2013 12:36:06 GMT
Call-ID: 273684113831d61a4520dcff39b44004@217.75.206.97
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 452

v=0
o=CiscoSystemsSIP-GW-UserAgent 4046 1496 IN IP4 217.75.205.49
s=SIP Call
c=IN IP4 217.75.205.49
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15
c=IN IP4 217.75.205.49
m=image 0 udptl t38
c=IN IP4 217.75.205.49
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

Re: No path to translate from SIP/logosoft-00000003(256) to

PostPosted: Mon Jan 14, 2013 6:46 am
by DomeDan
This is a codec translation issue, seams like your getting g729 data from your provider but you dont got the codec to translate it to ulaw
(if your codecs id is the same as mine)
Code: Select all
vicibox*CLI> show codecs
Disclaimer: this command is for informational purposes only.
   It does not indicate anything about your configuration.
        INT    BINARY        HEX   TYPE       NAME   DESC
--------------------------------------------------------------------------------
          1 (1 <<  0)      (0x1)  audio       g723   (G.723.1)
          2 (1 <<  1)      (0x2)  audio        gsm   (GSM)
          4 (1 <<  2)      (0x4)  audio       ulaw   (G.711 u-law)
          8 (1 <<  3)      (0x8)  audio       alaw   (G.711 A-law)
         16 (1 <<  4)     (0x10)  audio   g726aal2   (G.726 AAL2)
         32 (1 <<  5)     (0x20)  audio      adpcm   (ADPCM)
         64 (1 <<  6)     (0x40)  audio       slin   (16 bit Signed Linear PCM)
        128 (1 <<  7)     (0x80)  audio      lpc10   (LPC10)
        256 (1 <<  8)    (0x100)  audio       g729   (G.729A)
        512 (1 <<  9)    (0x200)  audio      speex   (SpeeX)
       1024 (1 << 10)    (0x400)  audio       ilbc   (iLBC)
       2048 (1 << 11)    (0x800)  audio       g726   (G.726 RFC3551)
       4096 (1 << 12)   (0x1000)  audio       g722   (G722)
      65536 (1 << 16)  (0x10000)  image       jpeg   (JPEG image)
     131072 (1 << 17)  (0x20000)  image        png   (PNG image)
     262144 (1 << 18)  (0x40000)  video       h261   (H.261 Video)
     524288 (1 << 19)  (0x80000)  video       h263   (H.263 Video)
    1048576 (1 << 20) (0x100000)  video      h263p   (H.263+ Video)
    2097152 (1 << 21) (0x200000)  video       h264   (H.264 Video)

remove "allow=g729" from your account entry and see if works and the carrier sends you something you can translate,
run "sip show peer logosoft" to view move info about the connection to the carrier

Re: No path to translate from SIP/logosoft-00000003(256) to

PostPosted: Tue Jan 15, 2013 10:10 am
by ctc_olsen
I think this is a dialplan entry issue. Looks like you need a prefix that needs to be set. Check our current config.

Code: Select all
exten => _981XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _981XXXXXXXXXX,2,Dial(${SIPTRUNKE}/${EXTEN:2},60,tTor)
exten => _981XXXXXXXXXX,3,Hangup


When you check the campaign for this, we set 98 as the dial prefix. Hope this helps or least point you to the right direction.
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