thr3ads.net - asterisk users - [asterisk-users] outgoing calls [May 2007]

If this information is useful, please help other people find it:
Share via:

Josu Lazkano Lete

2007-May-08 04:28 UTC

[asterisk-users] outgoing calls

Skipped content of type multipart/alternative-------------- next part
--------------
A non-text attachment was scrubbed...
Name: zaptel.conf
Type: application/octet-stream
Size: 34 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/f4c36cab/zaptel.obj
-------------- next part --------------
A non-text attachment was scrubbed...
Name: extensions.conf
Type: application/octet-stream
Size: 464 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/f4c36cab/extensions.obj
-------------- next part --------------
A non-text attachment was scrubbed...
Name: sip.conf
Type: application/octet-stream
Size: 252 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/f4c36cab/sip.obj
-------------- next part --------------
A non-text attachment was scrubbed...
Name: zapata.conf
Type: application/octet-stream
Size: 357 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/f4c36cab/zapata.obj

Josu Lazkano Lete

2007-May-08 04:35 UTC

head link

[asterisk-users] outgoing calls

Skipped content of type multipart/alternative-------------- next part
--------------
A non-text attachment was scrubbed...
Name: asterisk ona3.rar
Type: application/octet-stream
Size: 981 bytes
Desc: not available
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/16d5589f/asteriskona3.obj

Josu Lazkano Lete

2007-May-08 04:51 UTC

head link

[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@101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
From: 101 <sip:101@101>;tag=3159122210
To: "943833473" <sip:943833473@101>
Call-ID: 2591723875-241385386@10.0.0.9
CSeq: 1 INVITE
Contact: <sip:101@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 - 2591723875-241385386@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@101>;tag=3159122210
To: "943833473" <sip:943833473@101>;tag=as705b7b72
Call-ID: 2591723875-241385386@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 '2591723875-241385386@10.0.0.9' in 15000
ms
Found user '101'

<-- SIP read from 10.0.0.9:5060:
ACK sip:943833473@101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK105582920839628744;rport
From: 101 <sip:101@101>;tag=3159122210
To: "943833473" <sip:943833473@101>;tag=as705b7b72
Call-ID: 2591723875-241385386@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@101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
From: 101 <sip:101@101>;tag=3159122210
To: "943833473" <sip:943833473@101>
Call-ID: 2591723875-241385386@10.0.0.9
CSeq: 2 INVITE
Contact: <sip:101@10.0.0.9:5060>
Proxy-Authorization: Digest username="101",
realm="asterisk", nonce="5a68d228",
uri="sip:943833473@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 - 2591723875-241385386@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@101>;tag=3159122210
To: "943833473" <sip:943833473@101>;tag=as705b7b72
Call-ID: 2591723875-241385386@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@101 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.9:5060;branch=z9hG4bK22917179002522020113;rport
From: 101 <sip:101@101>;tag=3159122210
To: "943833473" <sip:943833473@101>;tag=as705b7b72
Call-ID: 2591723875-241385386@10.0.0.9
CSeq: 2 ACK
Proxy-Authorization: Digest username="101",
realm="asterisk", nonce="5a68d228",
uri="sip:943833473@101",
response="0fa2c9d246b54d137046e23743e1951c", algorithm=MD5
max-forwards: 70
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '2591723875-241385386@10.0.0.9'
Destroying call '44721582-00111639@10.0.0.9'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.0.0.9:5060:
OPTIONS sip:101@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@10.0.0.7>;tag=as0cc11f28
To: <sip:101@10.0.0.9:5060>
Contact: <sip:asterisk@10.0.0.7>
Call-ID: 5e0885a1360422511624720e14422af0@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@10.0.0.7>;tag=as0cc11f28
To: <sip:101@10.0.0.9:5060>
Call-ID: 5e0885a1360422511624720e14422af0@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 '5e0885a1360422511624720e14422af0@10.0.0.7'

here are my configuration files:

zapata.conf:

[channels]

signalling=fxs_ks

usecallerid=yes

callwaiting=no

threewaycalling=no

transfer=yes

cancallforward=yes

echocancel=yes

echotraining=yes

echocancelwhenbridged=no

rxgain=0

txgain=0

group=1

callgroup=1

pickupgroup=1

immediate=no

faxdetect=incoming

answeronpolarityswitch=yes

hanguponpolarityswitch=yes

polarityonanswerdelay=600

progzone=es

channel => 1



zaptel.conf:

loadzone=es

defaultzone=es

fxsks=1



extensions.conf:

[general]

static=yes

writeprotect=yes

[SOME]

exten => 101,1,Dial(SIP/101,30,Ttm)

exten => 101,2,Hangup

exten => 102,1,Dial(SIP/102,30,Ttm)

exten => 102,2,Hangup

[incoming]

exten => s,1,Wait(1)

exten => s,2,Answer()

exten => s,3,Dial(SIP/101,30,Ttm)

[outgoing]

exten =>_9XXXXXXXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)

exten =>_9XXXXXXXX,2,Hangup()

exten =>_9XXXXXXXX,102,Hangup()

[default]

exten => s,1,Answer()

exten => s,2,Wait(1)

exten => s,3,Dial(SIP/101,30,Ttm)


Thanks for all
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070508/8d4931f5/attachment-0001.htm

Apparently Analagous Threads

Search for more apparently analagous threads

asterisk users - May 2007 - outgoing calls

[asterisk-users] outgoing calls

[asterisk-users] outgoing calls

[asterisk-users] outgoing calls

Apparently Analagous Threads