[Asterisk-Users] Sound issue with Asterisk

2006-01-19 Thread Kevin

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
zaptel192516  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:[EMAIL PROTECTED];tag=as4a36d77b
To: sip:[EMAIL PROTECTED];tag=a0efbf44ecab5900
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Grandstream BT100 1.0.6.7
Contact: sip:[EMAIL PROTECTED]
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'

-- SIP read from 64.7.189.14:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
Call-ID: [EMAIL PROTECTED]
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 - [EMAIL PROTECTED]
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:[EMAIL PROTECTED];tag=b72941c93fe74588
To: sip:[EMAIL PROTECTED];tag=as6f00184d
Call-ID: [EMAIL PROTECTED]
CSeq: 19606 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=351ca5f6
Content-Length: 0


---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms

-- SIP read from 64.7.189.14:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c
From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588
To: sip:[EMAIL PROTECTED];tag=as6f00184d
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7
From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
Proxy-Authorization: Digest username=budgeTone-PubIP, 
realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], 
nonce=351ca5f6, response=3748b6120c7f4ecc4873cbdaf178d507

Call-ID: [EMAIL PROTECTED]
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 - [EMAIL PROTECTED]
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 

Re:[Asterisk-Users] Sound issue with Asterisk

2006-01-19 Thread Kevin

Hi everyone and Steve,

Well the problem I wrote about is fixed. Here is what I did to resolve 
the issue.


I was running kernel 2.6.11-1.1369_FC4smp before. I went and upgraded to 
2.6.14-1.1656_FC4smp along with the development files (which I finally 
found were not installed). After I installed the new kernel, recompiled 
asterisk, added my SIP and Extensions back in and all the menus in the 
demo, echo, etc, were all working.


Thanks Steve for helping me and hopefully this will help someone out as 
well.


Kevin

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