[Asterisk-Users] Sound issue with Asterisk
Kevin
kevin at locker70.com
Thu Jan 19 10:45:50 MST 2006
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
======================
Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
anything in there that related to what you said. Or is it in a
different file or location?
I am in the office now so I am able to provide some more information
about the issue that I am having.
Here is the kernel if this helps Fedora core 4 -- 2.6.11-1.1369_FC4smp
I know that ztdummy is at least loaded now. Also as stated before there
is nothing plugged into the T1 card. So I wasn't sure if that was
causing a problem or not which is why I enabled ztdummy but it was not
the first time I e-mailed you.
# lsmod | grep ztdummy
ztdummy 7748 0
zaptel 192516 6 ztdummy,wct4xxp
If I look at the connections from tcpdump, I see my phone call coming
in, but no traffic is being sent back to the phone. With an Echo() test,
I see the traffic going back and forth, but when I call into a menu,
then there is nothing.
Thanks,
Kevin
I ran a sip debug as well but I felt it was better at the end of the
e-mail:
<-- SIP read from 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK2b87d1ab;rport
From: "asterisk" <sip:asterisk at 64.7.161.26>;tag=as4a36d77b
To: <sip:budgeTone-PubIP at 64.7.189.14>;tag=a0efbf44ecab5900
Call-ID: 286e70e60596cc1b34a1fcac4e4c5337 at 64.7.161.26
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.6.7
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '286e70e60596cc1b34a1fcac4e4c5337 at 64.7.161.26'
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Supported: replaces
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (13 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360 at 64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKa4b619218b6ad43c;received=64.7.189.14
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as6f00184d
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Proxy-Authenticate: Digest realm="asterisk", nonce="351ca5f6"
Content-Length: 0
---
Scheduling destruction of call '2143389df4877360 at 64.7.189.14' in 15000 ms
<-- SIP read from 64.7.189.14:5060:
ACK sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as6f00184d
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19606 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
INVITE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Supported: replaces
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="3748b6120c7f4ecc4873cbdaf178d507"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 337
v=0
o=budgeTone-PubIP 8000 8001 IN IP4 64.7.189.14
s=SIP Call
c=IN IP4 64.7.189.14
t=0 0
m=audio 5004 RTP/AVP 2 8 4 18 15 97 9
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:15 G728/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=ptime:20
--- (14 headers 16 lines)---
Using INVITE request as basis request - 2143389df4877360 at 64.7.189.14
Sending to 64.7.189.14 : 5060 (non-NAT)
Found peer 'budgeTone-PubIP'
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 15
Found RTP audio format 97
Found RTP audio format 9
Peer audio RTP is at port 64.7.189.14:5004
Found description format G726-32
Found description format PCMA
Found description format G723
Found description format G729
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x519
(g723|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x400 (ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 500 in local (domain 64.7.161.26)
list_route: hop: <sip:budgeTone-PubIP at 64.7.189.14>
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Length: 0
---
-- Executing Playback("SIP/budgeTone-PubIP-7e44", "demo-abouttotry")
in new stack
We're at 64.7.161.26 port 13648
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Reliably Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 3026 3026 IN IP4 64.7.161.26
s=session
c=IN IP4 64.7.161.26
t=0 0
m=audio 13648 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -
---
-- Playing 'demo-abouttotry' (language 'en')
<-- SIP read from 64.7.189.14:5060:
ACK sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKb9dc124752ee52c9
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Contact: <sip:budgeTone-PubIP at 64.7.189.14>
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="eb1f3ad109804e62f1ba1edcc3706bf9"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19607 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (12 headers 0 lines)---
<-- SIP read from 64.7.189.14:5060:
BYE sip:500 at 64.7.161.26 SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKd592db0d7856fb18
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Proxy-Authorization: Digest username="budgeTone-PubIP",
realm="asterisk", algorithm=MD5, uri="sip:500 at 64.7.161.26",
nonce="351ca5f6", response="677ef75492337c32f50f3a936f4c0919"
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19608 BYE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
--- (11 headers 0 lines)---
Sending to 64.7.189.14 : 5060 (non-NAT)
Transmitting (no NAT) to 64.7.189.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
64.7.189.14;branch=z9hG4bKd592db0d7856fb18;received=64.7.189.14
From: "Budge Tone" <sip:budgeTone-PubIP at 64.7.161.26>;tag=b72941c93fe74588
To: <sip:500 at 64.7.161.26>;tag=as324cfd6f
Call-ID: 2143389df4877360 at 64.7.189.14
CSeq: 19608 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:500 at 64.7.161.26>
Content-Length: 0
---
== Spawn extension (local, 500, 1) exited non-zero on
'SIP/budgeTone-PubIP-7e44'
Destroying call '2143389df4877360 at 64.7.189.14'
More information about the asterisk-users
mailing list