RE: [Asterisk-Users] app_prepaid NAT issue
Because whenever I place call from the phone without it going through the
prepaid application, I don't have any audio issues and the route between the
gateway and the phone is built correctly with NAT taken into account. When
the call goes through app_prepaid, a new dial command is issued via
app_prepaid:
line 567:
res = prepaid_pbx_dial(chan, dialstr);
which then calls this:
line 369:
app = pbx_findapp(Dial);
if (app) {
ret = pbx_exec(chan, app, data, 1);
} else {
ast_log(LOG_WARNING, Could not find application (Dial)\n);
ret = -2;
}
I am assuming that this is where the issue is, but I am not famaliar enough
with the rest of the Asterisk code to know where to easily look next for
pbx_exec or what is required for NAT proxy to function correctly. I was
hoping that someone may have experienced something similar or could direct
me in the right direction.
Thanks,
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian K. West
Sent: Friday, June 18, 2004 8:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] app_prepaid NAT issue
Its not an apps place to take nat int account. WHERE did you get the idea
that it was?
bkw
- Original Message -
From: Brian Rathman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 18, 2004 5:03 PM
Subject: [Asterisk-Users] app_prepaid NAT issue
I was able to get app_prepaid working, but unfortunately I am getting one
way audio on the phone that I was placing the call from. It is behind NAT.
It appears that the app_prepaid is not taking this into consideration
since
I see:
Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route:
build_route:
Contact hop: sip:[EMAIL PROTECTED]:5060;line=jet7pbic
Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1406 ast_rtp_bridge: Oooh,
'SIP/7708183799-8d6d' changed end address to 192.168.1.101:10094 (format
6)
Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1408 ast_rtp_bridge: Oooh,
'SIP/7708183799-8d6d' was 65.202.115.115:10094/(format 6)
Any help would be greatly appreciated.
Thanks,
Brian
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[Asterisk-Users] app_prepaid NAT issue
I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=jet7pbic Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1406 ast_rtp_bridge: Oooh, 'SIP/7708183799-8d6d' changed end address to 192.168.1.101:10094 (format 6) Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1408 ast_rtp_bridge: Oooh, 'SIP/7708183799-8d6d' was 65.202.115.115:10094/(format 6) Any help would be greatly appreciated. Thanks, Brian ___ 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] NAT and Qualify Question
I have several SNOM200 phones at various remote locations all behind some kind of NAT. Unfortunately I see where most of the phones go from REACHABLE to UNREACHABLE quite often. Is there anything that I can change to help with this issue. Length of registration? qualify time? Any help would be greatly appreciated. Thanks, Brian ___ 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] SIP Registration Problem
Title: Re: [Asterisk-Users] Wiki TOS - worrying for an open sourceproject? I am using snom200 phones registering with Asterisk via SIP. I can see where the phone registers without a problem, and then when you try and make a call I get a proxy authentication required message on the phone and failed to authenticate user error in the Asterisk messages file. Then the next call you make from the phone goes through without a problem. Nothing changes between these two events, but it is almost like the phone is using two different passwords for the same account. Has anyone else seen a problem like this? I am using an Asterisk CVS version from early March, not sure if upgrading will help as well. Thanks, Brian
[Asterisk-Users] Three Way Calling/Conferencing
Title: RE: [Asterisk-Users] Dial Plan Format Strings
I
currently have about 40 users up on Asterisk and it is working great. One issue
I have though is the inability to conference calls/3-way callingon my SNOM
200 phones. Whenever I press theCNF/TRAN button on the phone, it just
drops the current call.I noticed a testimonial on the Wiki about doing
this with Asterisk and the SNOMs, but I can not seem to get it to work. Is
anyone out there doing this and if so can you point me in the right
direction?
Thanks,
Brian
-Original Message-From: Benjamin Wakefield
[mailto:[EMAIL PROTECTED]Sent: Tuesday, April 13, 2004 10:29
AMTo: [EMAIL PROTECTED]Subject: RE:
[Asterisk-Users] Dial Plan Format Strings
snip have to dial the entire
number, like 1 + area code + number. I'd like to eliminate this by having the user just dial 9 + 7 digit number, and
have asterisk put the 1 + area code (which is in a
variable in extensions.conf) in front of it prior to sending the request to Voice Pulse. Is
this possible? /snip
Sure it's possible! Asterisk can do anything!
exten =
_9XXX,1,Dial(Technology/123/1212${EXTEN:1})
See after the: "Technology/123/" there is a "1212" you can
make that your "1 + area code"
Then the: "${EXTEN:1}" dumps in the number that was dialled
and chops off the first (1) digit, which is the
9.
:) Ben Benjamin
Wakefield [EMAIL PROTECTED] http://www.dcsi.net.au/
DCSI - We do Internet. 64 Queen
Street Warragul, VIC 3820 AU Ph: (+61) 1300 665 575 Fx: (+61) 1300 556
595
-BEGIN GEEK CODE BLOCK- Version: 3.12 G! d- s: a-- C+ UL++ P+ L++ E
W+ N+ o- K- w+$ O--- M-- V? PS !PE Y-- PGP- t 5 X+ R-
tv b- DI-- D--- G-- e* h* r- z++ --END GEEK CODE
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[Asterisk-Users] Passing DTMF
Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco AS5300 with * in the media stream. Unfortunately, the only way I can get the calls to connect is with t or T at the end of the Dial() statement and then that picks off the dtmf digits. I have tried the canreinvite=yes on both the phone peer and the gateway peer and I still have to add the T to the Dial statement to make the call complete. Any suggestions??? Thanks, Brian ___ 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] Passing DTMF
Title: [Asterisk-Users] Passing DTMF Just to follow up, it does not matter what codec I use, and when I listen to the call on the far end, I can hear a very quick blip that sounds like the correct tone, but it is not long enough for an IVR to recognize. Is there a way to boost the length of this tone in Asterisk? Any help would be greatly appreciated. -Original Message-From: Brian J. Rathman [mailto:[EMAIL PROTECTED]Sent: Tuesday, April 06, 2004 1:29 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Passing DTMF Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco AS5300 with * in the media stream. Unfortunately, the only way I can get the calls to connect is with t or T at the end of the Dial() statement and then that picks off the dtmf digits. I have tried the canreinvite=yes on both the phone peer and the gateway peer and I still have to add the T to the Dial statement to make the call complete. Any suggestions??? Thanks, Brian ___ 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] Passing DTMF
Title: Re: [Asterisk-Users] Passing DTMF I have tried every combination of codec and dtmfmode. I can hear the dtmf tone on the far end phone, it just appears to be to short. Is there a way to increase the duration of the tone? -Original Message-From: Eric Wieling [mailto:[EMAIL PROTECTED]Sent: Tuesday, April 06, 2004 3:10 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Passing DTMF On Tue, 2004-04-06 at 12:29, Brian Rathman wrote: Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco AS5300 with * in the media stream. Unfortunately, the only way I can get the calls to connect is with t or T at the end of the Dial() statement and then that picks off the dtmf digits. I have tried the canreinvite=yes on both the phone peer and the gateway peer and I still have to add the T to the Dial statement to make the call complete. Any suggestions??? cantrinvite=yes tells asterisk to, if it can, remove itself from the media stream. T and t and r and many other Dial options tells Asterisk to stay in the media stream so it can listen to the DTMF. None of this has ANYTHING to do with passing DTMF between the two endpoints (except of course passing # for t or T). If you cannot pass DTMF between the two endpoints then something ELSE is wrong. Maybe you are trying to use inband DTMF with a compressed codec. Inband DTMF will only work with ulaw or alaw codecs. --Eric -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ 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] DTMF Passing
I am trying to get dtmf digits to pass from a SNOM 200 through * to a Cisco AS5300. I have setup the cisco gateway and the sip.conf file to use rfc2833 and I have disabled inband dtmf on the snom 200. Unfortunately, the digits are still not being passed. Something tells me that I am missing something in the extensions.conf file, but I am at a loss. I would greatly appreciate any help you can give me. Thanks, Brian ___ 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] Auto connect to voicemail
I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables and macros for the extensions file. Can someone please help me with this issue? Thanks, Brian ___ 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] Extensions.conf sending calls to Cisco AS5300
I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought that the T had something to do with a timeout, but I have also seen documentation referencing that it allows * to stay in the middle of the call to determine if the customer use the # key, etc. I have not been able to find the detailed documentation that I was looking for on this subject. Can someone please direct me to this? Also it is my understanding, that if * stays in the middle of the call, I can not use the g729 codec without licensing from Digium. If this is the case, is there a way that I can use g729 in pass thru and still complete calls to the gateway? Any help would be greatly appreciated. Thanks, Brian ___ 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] SIP Calls with * and Cisco AS5300
I am attempting to send calls from various Ip phones (Snom 105,200 and SIP Express ATA) to a default SIP gateway (Cisco AS5300) and for some reason my calls are failing 2 seconds after the called party picks up. This is what the Asterisk console is displaying during the call attempt: -- Executing Dial(SIP/7708183797-28a7, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- SIP/66.102.15.212-bfdc is making progress passing it to SIP/7708183797-28a7 -- SIP/66.102.15.212-bfdc answered SIP/7708183797-28a7 -- Attempting native bridge of SIP/7708183797-28a7 and SIP/66.102.15.212-bfdc -- Got SIP response 481 Invalid CSeq Number back from 66.102.15.212 == Spawn extension (default, 6783528833, 1) exited non-zero on 'SIP/7708183797-28a7' At this point I am not sure if it is a problem with the config on my cisco box, or with the setup in my extension or sip.conf files. The cisco box is handling SIP calls from another registrar now, so I doubt that it is the problem. Has anyone been able to get this setup working? Thanks, Brian ___ 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
