stotaro at totarotechn...
Guest
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 Posted: Fri Mar 21, 2008 8:26 am Post subject: [asterisk-users] Newbie: Two problems with Asterisk Config, |
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Probably a codec issue. SIP debug while making a call would be helpful.
Thanks,
Steve Totaro
On Fri, Mar 21, 2008 at 4:06 AM, Pete Kay <petedao at gmail.com> wrote:
| Quote: | Hi,
I switched to Wengo and solved the one "beat"problem. However, I am still
not able to listen to the recorded .wav sound. Can anyone please point me
to the right direction? How to listen to the .wav sound?
Thanks,
Pete
On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas <crt.rojas at gmail.com> wrote:
| Quote: | Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay <petedao at gmail.com> wrote:
| Quote: |
Hi,
I am sorry my questinos are too fundamental. I am new to Asterisk, and
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hope to catch up as fast as I can.
| Quote: | | Quote: |
Problem 1:
I have my SIP client ( in one PC .102) and SIP server ( in another PC
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.101) within the same land. They can make SIP connection, but when the SIP
client makes call to play an audio file, I can only hear a "beat" sounds,
and then nothing else. In the console, I can see:
| Quote: | | Quote: | *CLI> -- Executing [111 at my-phones:1] Answer("SIP/2001-081dd6e0", "")
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in new stack
| Quote: | | Quote: | -- Executing [111 at my-phones:2] VoiceMail("SIP/2001-081dd6e0",
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"2000") in new stack
| Quote: | | Quote: | Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts
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000160, len 000160)
| Quote: | | Quote: | -- <SIP/2001-081dd6e0> Playing 'vm-intro' (language 'en')
Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts
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000320, len 000160)
| Quote: | | Quote: | Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts
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000480, len 000160)
| Quote: | | Quote: | Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037721, ts
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000640, len 000160)
| Quote: | | Quote: | Got RTP packet from 192.168.1.102:8000 (type 00, seq 062222, ts
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1373137124, len 000160)
| Quote: | | Quote: | Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037722, ts
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000800, len 000160)
| Quote: | | Quote: | Sent RTP packet to 192.168.1.102:8000 (type 00, seq 037723, ts
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000960, len 000160)
| Quote: | | Quote: |
Is it the prolem? First it sends to the public address of the the
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router, then it sends to the virtual IP. Is this the problem that causing
my to hear just one "beat" sound and then no audio?
| Quote: | | Quote: |
Problem 2:
The problem is isolated from Problem 1, cuz I run the SIP client on the
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same machine as the server, so there should not be network problem. I
recorded some voice mails and they are stored as .wav files ok. When I
tried to hear back the message, It does not work. Is there any
configuration that I have to go through to have Asterisk to play .wav file?
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