Problem of SIP and many other Things

Hi Folks,:roll:

I’m here got tired of , when I’m using asterisk on remote server.

First of all,
Does [color=red]Cepstral really work with asterisk?[/color] … If yes in my case it is not accepting digits pressed for menus provided by Agi
I’m providing menus such as…
Press 1, for accessing your voice mail account.
press 2, for deleting your messages.

when user presses 1 my asterisk does not take that choice.
Does this is the problem of Cepstral or asterisk can’t work with remote machines?

I did not find information regarding [color=red]Installing Digium TDM2400P
anywhere on any site[/color].
Could anybody please provide me that?

After this all happened to me, I removed asterisk formatted machine and again installed asterisk. after that when I dial from my remote X-Lite
It’s Giving me following error.

[Mar 6 16:28:49] WARNING[29689]: chan_sip.c:1947 retrans_pkt: Maximum retries exceeded on transmission NzY5Y2YyZDM5Y2U4NDVhZTgyMTczZWI5YTRlMWE5Nzk. for seqno 2 (Critical Response)

and I cant here a simple playback message.
I this particular case I have added in extensions.conf following lines.

exten => 123,1,Answer()
exten => 123,2,Festival(Asterisk and festival are working together)
exten => 123,3,Hangup()

also in above case I have tried with,

exten => 123,1,Answer()
exten => 123,2,Playback(you-sound-cute)
exten => 123,3,Hangup()

I’m not able to here anything when I dial exten 123 from sip.[even “YOU SOUND CUTE” also]
just my call goes on and on and I have to manually Hangup from X-Lite.
and just get following message.

[Mar 6 16:29:05] WARNING[29689]: chan_sip.c:1947 retrans_pkt: Maximum retries exceeded on transmission MDFmMmY4NTZhMDEwYzg4ZmQ0Nzg2NGM0OGMwZGQyYzY. for seqno 2 (Critical Response)
:confused: :confused:
Note that, My asterisk is Remote one and I have a SSH access for that machine. by which I have configured asterisk on that machine.

Thanks and please help me out in this,
Rahul Borkar. :frowning: :frowning: :frowning: :frowning: :frowning: :frowning: :frowning: :frowning: :frowning: :frowning:

Hi again…

I got one thing by which I’m getting sound at X-Lite but this Works only for PlayBack and Background application in asterisk and Not for festival and cepstral [I’m not getting any voices played at X-Lite, whilw asterisk server shows that it is playing voices].

Because of following thing my [color=red]Playback application[/color] at remote asterisk started to work. Even I have that maximum_sip_retransmit warning

  1. under sip.conf I gave option of [color=red]qualify=route[/color]
  2. I added stun for x-lite softphone.

[color=red]But my new problem is , this is not working for Cepstral and festival?[/color]
Please help me out in this. I’m really searching for solution badly.

Thanks and regards,
Rahul R .Borkar

Please help me out in this

Hi all…

[size=200][color=orange]I’m searching Solution badly please help me out in this…?[/color][/size]

if you want to make an IVR why you are using “festival” ?
1- sip show peers…?
2- did you type the same dialplan in the same context ?


  1. I’m using Festival as a first step. In second step I’m going to install Cepstral TTS.

  2. When I typed command sip show peers it shows me following things.

Name/username Host Dyn Nat ACL Port Status
102/102 D N 48927 OK (528 ms)
101/101 D N 27862 UNREACHABLE
2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]

  1. Now about dial Plan and contexts I created a context as sip in extensions.conf and given same as context in sip.conf under both of those users 101 and 102 so that when they call they’ll jump to context sip under extensions.conf.

Problem is I’m getting sounds if I play a readymade GSM file with the help of Playback function but I’m not able to hear sound in x-Lite if I use TTS to say something.
Eg: if i use
exten => 123,1,Playback(you-sound-cute) or SayNumber(1234) it works fine but if I use,
exten =>123,1,Answer()
exten =>123,2,Festival(hi all Hows Life)
I’m not able to hear anything at X-Lite.

Note that asterisk is running at remote site and I have an IP for connecting x-Lite to it.
and also I’m using asterisk 1.4.18 and all other simmillar packages.

Rahul Borkar

did you configure correctly X-Lite (options > devices ?
did the “gsm” Codec is installed ?
when you make a call with asterisk, you can hear something?

I can’t check with sounds as asterisk is on remote machine and I’m facing problem on that machine only. My local asterisk server with same environment works fine with same setup but when I reflected that setup to remote asterisk server then it started giving me problems.

For allowing codecs I have allowed all codecs under x-Lite and it has shown all codes under enabled codecs ListBox.

I’m not getting what the problem is with TTS as my gsm file gets played successfully but my TTS message does not get played