[asterisk-users] outgoing calls

Tue May 8 04:53:26 MST 2007

Hello Josu,

In you're sip.conf you have the 2 phones configured that they are in the SOME context.

Looking at the SOME contect in extensions.conf you only have the 2 phones defined. If you want to call ouside from the SOME context as well, you need to include the outgoing context there as well.

Regards, 

Roelof Dijkstra 
Network Engineer EMEA 
Compuware Europe BV 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Josu Lazkano Lete
Sent: Tuesday, May 08, 2007 1:36 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] outgoing calls

hello friends, I have a problem when I call to outside (9XXXXXXXX) from IPs Telephones.

the incomning calls are OK.

in the console when I put "sip debug peer 101" I have this lines:

*CLI> sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI>
<-- SIP read from 10.0.0.9:5060:
INVITE sip:943833473 at 101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 1 INVITE
Contact: <sip:101 at 10.0.0.9:5060>
max-forwards: 70
supported: 100rel
user-agent:
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 278

v=0
o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9
s=A conversation
c=IN IP4 10.0.0.9
t=0 0
m=audio 10010 RTP/AVP 18 4 4 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv

--- (13 headers 13 lines) ---
Using INVITE request as basis request -  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
Sending to 10.0.0.9 : 5060 (NAT)
Reliably Transmitting (no NAT) to 10.0.0.9:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;received=10.0.0.9;rport=5060
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5a68d228"
Content-Length: 0

---
Scheduling destruction of call  <mailto:'2591723875-241385386 at 10.0.0.9'> '2591723875-241385386 at 10.0.0.9' in 15000 ms
Found user '101'

<-- SIP read from 10.0.0.9:5060:
ACK sip:943833473 at 101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 1 ACK
max-forwards: 70
Content-Length: 0

--- (8 headers 0 lines) ---

<-- SIP read from 10.0.0.9:5060:
INVITE sip:943833473 at 101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 2 INVITE
Contact: <sip:101 at 10.0.0.9:5060>
Proxy-Authorization: Digest username="101", realm="asterisk", nonce="5a68d228", uri="sip:943833473 at 101", response="0fa2c9d246b54d137046e23743e1951c", algorithm=MD5
max-forwards: 70
supported: 100rel
user-agent:
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 278

v=0
o=sdp_admin 11288299 31318536 IN IP4 10.0.0.9
s=A conversation
c=IN IP4 10.0.0.9
t=0 0
m=audio 10010 RTP/AVP 18 4 4 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendrecv

--- (14 headers 13 lines) ---
Using INVITE request as basis request -  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
Sending to 10.0.0.9 : 5060 (NAT)
Found user '101'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Peer audio RTP is at port 10.0.0.9:10010
Found description format G729
Found description format G723
Found description format G723high
Found description format PCMA
Found description format PCMU
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 943833473 in SOME (domain 101)
Reliably Transmitting (no NAT) to 10.0.0.9:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;received=10.0.0.9;rport=5060
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---

<-- SIP read from 10.0.0.9:5060:
ACK sip:943833473 at 101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
From: 101 <sip:101 at 101>;tag=3159122210
To: "943833473" <sip:943833473 at 101>;tag=as705b7b72
Call-ID:  <mailto:2591723875-241385386 at 10.0.0.9> 2591723875-241385386 at 10.0.0.9
CSeq: 2 ACK
Proxy-Authorization: Digest username="101", realm="asterisk", nonce="5a68d228", uri="sip:943833473 at 101", response="0fa2c9d246b54d137046e23743e1951c", algorithm=MD5
max-forwards: 70
Content-Length: 0

--- (9 headers 0 lines) ---
Destroying call  <mailto:'2591723875-241385386 at 10.0.0.9'> '2591723875-241385386 at 10.0.0.9'
Destroying call  <mailto:'44721582-00111639 at 10.0.0.9'> '44721582-00111639 at 10.0.0.9'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.9:5060:
OPTIONS sip:101 at 10.0.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport
From: "asterisk" <sip:asterisk at 10.0.0.7>;tag=as0cc11f28
To: <sip:101 at 10.0.0.9:5060>
Contact: <sip:asterisk at 10.0.0.7>
Call-ID:  <mailto:5e0885a1360422511624720e14422af0 at 10.0.0.7> 5e0885a1360422511624720e14422af0 at 10.0.0.7
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 May 2007 11:34:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---

<-- SIP read from 10.0.0.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.7:5060;branch=z9hG4bK65c2f229;rport
From: "asterisk" <sip:asterisk at 10.0.0.7>;tag=as0cc11f28
To: <sip:101 at 10.0.0.9:5060>
Call-ID:  <mailto:5e0885a1360422511624720e14422af0 at 10.0.0.7> 5e0885a1360422511624720e14422af0 at 10.0.0.7
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

--- (8 headers 0 lines) ---
Destroying call  <mailto:'5e0885a1360422511624720e14422af0 at 10.0.0.7'> '5e0885a1360422511624720e14422af0 at 10.0.0.7'

I attached my configuration files.

Thanks for all

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