Chain_sip route to incorrect DID

Asterisk Asterisk Support
1

I have 2 voip number configured correctly. I can make outbound call from both of them and receive call only from the first one. Seems that context DID(2) in users.conf route to DID(1) in extensions.conf. Please see the following configuration and debug list. What’s your opinion?

users.conf

[Eutelia1]
  context=DID_Eutelia1
  contact=03599999253
  host=voip.eutelia.it
  username=03599999253
  secret=xxxxxxxxxxxxx
  trunkname=Eutelia1
  hasiax=no
  registeriax=no
  hassip=yes
  registersip=yes
  trunkstyle=voip
  hasexten=no
  insecure=port,invite
  type=friend
  dtmfmode=auto
  realm=voip.eutelia.it
  fromdomain=voip.eutelia.it
  fromuser=03599999253
  disallow=all
  allow=ulaw,alaw,gsm,g726
  qualify=yes


[Eutelia2]
  context=DID_Eutelia2
  contact=03599999584
  host=voip.eutelia.it
  username=03599999584
  secret=xxxxxxxxxxxxx
  trunkname=Eutelia2
  hasiax=no
  registeriax=no
  hassip=yes
  registersip=yes
  trunkstyle=voip
  hasexten=no
  quilify=yes
  insecure=port,invite
  type=friend
  dtmfmode=auto
  realm=voip.eutelia.it
  fromdomain=voip.eutelia.it
  fromuser=03599999584
  disallow=all
  allow=ulaw,alaw,gsm,g726
  qualify=yes
extensions.conf

[DID_Eutelia1]
  include=DID_Eutelia1_default
  
[DID_Eutelia1_default]
  exten=03599999253,1,Goto(ringroups-custom-1,s,1)

[DID_Eutelia2]
  include=DID_Eutelia2_default
  
[DID_Eutelia2_default]
  exten=03599999584,1,Goto(ringroups-custom-1,s,1)
status

>sip show registry

Host                           dnsmgr Username       Refresh State                Reg.Time                 
voip.eutelia.it:5060           N      03599999253        315 Registered           Wed, 25 Mar 2009 16:50:48
voip.eutelia.it:5060           N      03599999584        315 Registered           Wed, 25 Mar 2009 16:50:48


>sip show peers

Name/username              Host            Dyn Nat ACL Port     Status     
601/601                    192.168.0.100    D   N      5088     OK (246 ms) 
603/603                    192.168.0.103    D   N      5060     OK (109 ms) 
611/611                    192.168.0.101    D   N      5083     OK (256 ms) 
Eutelia1/03599999253       83.211.227.21               5060     OK (80 ms) 
Eutelia2/03599999584       83.211.227.21               5060     OK (73 ms) 
********************************************
Verbosity 
calling from an external line to 03599999253
>>>> the calling is handled correctly
********************************************
  == Using SIP RTP CoS mark 5
    -- Executing [03599999253@DID_Eutelia1:1] Goto("SIP/03599999253-088cff20", "ringroups-custom-1,s,1") in new stack
    -- Goto (ringroups-custom-1,s,1)
    -- Executing [s@ringroups-custom-1:1] NoOp("SIP/03599999253-088cff20", "Office") in new stack
    -- Executing [s@ringroups-custom-1:2] Dial("SIP/03599999253-088cff20", "SIP/601&SIP/611&SIP/603&SIP/605,20,i") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 601
  == Using SIP RTP CoS mark 5
    -- Called 611
  == Using SIP RTP CoS mark 5
    -- Called 603





********************************************
Verbosity
calling from an esternal line to 03599999584
>>>> the calling stops for errors
********************************************
  == Using SIP RTP CoS mark 5
[Mar 25 17:37:26] NOTICE[3713]: chan_sip.c:18071 handle_request_invite: Call from '03599999253' to extension '03599999584' rejected because extension
not found.






********************************************
Debug
calling from an esternal line to 03599999584
>>>> the calling stops for errors
********************************************


<--- SIP read from UDP://83.211.227.21:5060 --->
INVITE sip:03599999584@60.61.62.63 SIP/2.0
Record-Route: <sip:83.211.227.21;ftag=D95E64FC-1D42;lr=on>
Record-Route: <sip:83.211.227.12;ftag=D95E64FC-1D42;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.12;branch=z9hG4bK67d5.b3b4b4d4.0
Via: SIP/2.0/UDP  62.94.88.138:5060;rport=61985;x-route-tag="tgrp:Slot6";branch=z9hG4bK126F591826
From: <sip:3386650106@62.94.88.138>;tag=D95E64FC-1D42
To: <sip:03599999584@voip.eutelia.it>
Call-ID: 972CFB10-189311DE-B2D88113-D9C02C99@62.94.88.138
CSeq: 102 INVITE
Max-Forwards:  8
Remote-Party-ID: <sip:3386650106@62.94.88.138>;party=calling;screen=yes;privacy=off
Contact: <sip:3386650106@62.94.88.138:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 415

v=0
o=CiscoSystemsSIP-GW-UserAgent 3606 6755 IN IP4 62.94.88.138
s=SIP Call
c=IN IP4 62.94.199.37
t=0 0
m=audio 64890 RTP/AVP 18 8 0 4 3 125 101
c=IN IP4 62.94.199.37
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=5.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:125 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (16 headers 17 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 83.211.227.21 : 5060 (no NAT)
Using INVITE request as basis request - 972CFB10-189311DE-B2D88113-D9C02C99@62.94.88.138
Found peer 'Eutelia1' for '3386650106' from 83.211.227.21:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 125
Found RTP audio format 101
Peer audio RTP is at port 62.94.199.37:64890
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format GSM for ID 3
Found unknown media description format X-CCD for ID 125
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gs
m|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 62.94.199.37:64890
[b]Looking for 03599999584 in DID_Eutelia1 (domain 60.61.62.63)[/b]

<--- Reliably Transmitting (no NAT) to 83.211.227.21:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21
Via: SIP/2.0/UDP 83.211.227.12;branch=z9hG4bK67d5.b3b4b4d4.0
Via: SIP/2.0/UDP  62.94.88.138:5060;rport=61985;x-route-tag="tgrp:Slot6";branch=z9hG4bK126F591826
From: <sip:3386650106@62.94.88.138>;tag=D95E64FC-1D42
To: <sip:03599999584@voip.eutelia.it>;tag=as39f0af51
Call-ID: 972CFB10-189311DE-B2D88113-D9C02C99@62.94.88.138
CSeq: 102 INVITE
Server: Asterisk_Eut
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 25 17:48:24] NOTICE[3713]: chan_sip.c:18071 handle_request_invite: Call from '03599999253' to extension '03599999584' rejected because extension
not found.
Scheduling destruction of SIP dialog '972CFB10-189311DE-B2D88113-D9C02C99@62.94.88.138' in 6400 ms (Method: INVITE)
company*CLI>
<--- SIP read from UDP://83.211.227.21:5060 --->
ACK sip:03599999584@60.61.62.63 SIP/2.0
Max-Forwards: 15
Record-Route: <sip:83.211.227.21;ftag=D95E64FC-1D42;lr=on>
Via: SIP/2.0/UDP 83.211.227.21;branch=0
Via: SIP/2.0/UDP 83.211.227.12;branch=z9hG4bK67d5.b3b4b4d4.0
From: <sip:3386650106@62.94.88.138>;tag=D95E64FC-1D42
Call-ID: 972CFB10-189311DE-B2D88113-D9C02C99@62.94.88.138
To: <sip:03599999584@voip.eutelia.it>;tag=as39f0af51
CSeq: 102 ACK
Content-Length: 0
2

Check my post in this thread: forums.digium.com/viewtopic.php? … ht=#126287, it should help you understand how to solve your problem.

Cheers.

Marco Bruni
www.marcobruni.net

3

Hello Marco,

Thank you a lot for your post. Your indications have resoved the drawback.
The only concern is this work around is not hadled by asterisk gui and the incoming call page does not work anymore.

I had a look to your site. Very nice! Good to know you’re Italian!

Thanks again

Bye

4

I’m sorry I can’t help on this, because I don’t use the Asterisk gui and I don’t know other workarounds.

Thanks.

Cheers.

Marco Bruni
www.marcobruni.net