[Asterisk-Users] Sipura line 1 outgoing voice problem?
Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8-- sip.conf --8-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_iax2.c:5309 socket_read: Received mini frame before first full voice frame
I'm trying to call my * box via iaxtel to diagnose my previous problem
(see earlier email today) and I get the following message spewing to my
console:
Mar 16 22:10:52 WARNING[9226]: chan_iax2.c:5309 socket_read: Received
mini frame before first full voice frame
It looks like this problem has been reported before (from July 2003 and
January 2004) but it has never been resolved. Is this a known issue or
am I doing something wrong? There shouldn't be any technical reasons
that I cannot call from * to the same * via iaxtel, right?
* version: Asterisk CVS-03/16/04-22:05:56 (just built to try to
alleviate this and another problem)
Relevent config files:
--8-- extensions.conf --8--
.
.
exten = _1700NXX,1,Macro(dialiaxtel,${EXTEN})
exten = _1700NXX,2,Congestion
.
.
[macro-dialiaxtel]
exten = s,1,Wait(1)
exten = s,2,SayDigits(5)
exten = s,3,Ringing
exten = s,4,Dial(IAX2/xxx:[EMAIL PROTECTED]/[EMAIL PROTECTED],,Tr)
exten = s,105,Noop
.
.
--8-- iax.conf --8--
.
.
register = xxx:[EMAIL PROTECTED]
.
.
[iaxtel]
type=user
context=default
auth=rsa
inkeys=iaxtel
-- Chris
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Re: [Asterisk-Users] chan_iax2.c:5309 socket_read: Received mini frame before first full voice frame
Adam Hart wrote: Could be simple packet loss, the first frame (the full frame) gets loss and retransmitted but in the mean time, the second frame (a mini frame) arrives. I guess that could be the case. My * box is behind NAT. I don't think that would reliably cause this issue though. I get the exact same result every time I try it, so I don't currently think its a timing problem with the reception of the packets. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_iax2.c:5309 socket_read: Received mini frame before first full voice frame
Mar 16 22:10:52 WARNING[9226]: chan_iax2.c:5309 socket_read: Received mini frame before first full voice frame Ok, I found the problem. The error message is not very descriptive but I was able to fix it by playing around. I had the register line in iax.conf below my peer definitions, and I seem to remember that it needs to be in the global section. I'm not sure I was successfully registering with iaxtel. I moved it to the global section and everything is working fine now (EXCEPT, I still have my original Sipura line 1 problem on incoming calls -- see previous email). Thanks for the help.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delta Three/iConnectHere Outgoing Caller ID?
Nate Carlson wrote: Caller ID to work. I searched the archives, and found some people saying that outgoing Caller ID shows up as Out of Area (that's what I get), and another person saying it worked 75% of the time for him. I've tried calling 3 different area codes (612, 952, and 253), so I've tried multiple My experience is that it worked 50% of the time for me. I live in 919. When I call to 919, it always comes up Unknown. When I call family in 304, the caller ID is set to my home number and it shows up as it should (both number and name) on my family's callerid units. I think the theory that some exchanges support it and some don't is solid. 1) For the people that have had caller ID working, what type of iConnectHere service plan do you have? (IE, do you have a number with them, or is it outgoing only?) Right now, I'm testing with the free $10 trial, outgoing only, no incoming number. 3) What's the proper way to configure things to get Caller ID to work, for the people that have it working? I'll include the configuration that I've tried below. Here is what I do on outgoing calls: exten = s,1,SetCallerID(919-XXX-) exten = s,2,SetCIDName(Your Name) Though, after reading other, more knowledgable people's explanation of how caller id numbers and names are linked, I think only the first line is necessary. FYI, these are the first two lines in my [macro-dialiconnect] macro. 5) Is my 'register' syntax below set up properly? I couldn't find much documentation on the 'proper' way to set this up. I don't think the /username is necessary at the end of the register command. To be sure, look at the log to see if * complains about the registration. I do know that if you don't have an incoming number, you don't need to register with iconnect at all. Just set up username/password in sip.conf and Dial(). -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does anyone manage the wiki?
I would like to correct some of the text on the GotoIf application page on the wiki. Does somebody actively manage changes like this, or should I fire away and make it myself? I'm actually surprised I have permission to edit a page without prior authorization, but it DOES state at the bottom of the generated pages to 'please update the page with new information...'. If I don't hear otherwise, I'll go ahead. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
Steven E. Frazier wrote: I have a similar set up, I don't have a separate sip phone, but I have the same exact problem with the line 1. I don't know if my config files aren't right, but I can't transfer between exts yet, but my issues is with line one on an incoming call from an X100P as well. FYI.. I wasn't able to transfer between analog phones on my sipura until I explicitly set the DTMF mode on both lines in the sipura web configuration to INFO (which matched what I already had in sip.conf). -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
