[asterisk-users] WirelessIP5000 SIP registration problem
I have a Hitachi WirelessIP5000 phone (the original 1st gen hardware). It's running the latest firmware (v2.2.6) and my Asterisk server is running 1.2.10. This setup has been working great for me for a long time, and then last week I started having a problem where the Hitachi phone loses registration. It registers fine when I first power it on, but at some point later the display shows Not Register and the phone never attempts to re-register after that. The length of time depends on the expire setting. If I press the Register soft key on the phone, it re-registers fine, only to lose registration again later on. I have not made any config changes to the phone. I have made some extensions.conf changes on the Asterisk server, but I can't think of any way that would cause this problem. I did look at a packet trace of the SIP communication between the phone and server. SIP registration expire is set to 60 seconds. When the phone first powers up, it sends a SIP REGISTER without an Authorization: header. The Asterisk server replies with 401 Unauthorized. The phone immediately sends another REGISTER with the Authorization: header included and succeeds. At 30 seconds after power up, the phone sends another REGISTER without Authorization: and gets 401 Unauthorized and then immediately retries, again without the Authorization: header. At 60 seconds after power up, the phone's display changes to Not Register and it never tries to register again unless I press the Register key or power cycle the phone. Anyone have any idea what could be going on here? I have tried downgrading the phone's firmware to v2.1.11 and had the same problem. I have not tried upgrading to a newer version of Asterisk, but seems like that shouldn't be necessary since thing were working fine before. Any help appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WIP-5000 and DTMF
My WIP-5000 phone is working well with my Asterisk box now, except for DTMF. All DTMF key presses come across as clipped or just clicks on the remote side. I had this problem with my Sipura ATA as well, but fixed that by playing with the settings on the Sipura device. I've tried dtmfmode=inband and also rfc2833, but neither seem to work. I don't see any place in the settings on the WIP-5000 to change its DTMF mode. Anyone have DTMF working with their WIP-5000 phone? Jim Meehan Oakland, CA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware
Good suggestion. It now seems to roam between access points nicely, even while a call is in progress. What access pooints are you using? One is an original Apple Airport (802.11b only) and the other is an Apple Airport Extreme (802.11b/g). -j ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware
Roman Volf wrote: Have you tried putting both access points on the same channel? Good suggestion. It now seems to roam between access points nicely, even while a call is in progress. Also, I found firmware v1.5.3 if anyone needs it, along with manuals that are quite a bit more in depth than the ones I had before. If you need the firmware or manauls, feel free to e-mail me off-list. -j ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware
I've got a Hitachi WIP-5000 phone. Seems to work well with my Asterisk setup, except for a few annoyances: 1) If the phone has been sitting unused for a while, and I dial an outbound call, it often fails. Doesn't matter what number I'm calling. If I redial, it always goes through fine, and all subsequent calls also go through fine, until it's been sitting around idle for a while again. Incoming calls always come through, regardless of how long it's been sitting. Maybe a NAT problem? I haven't started looking through SIP packet logs yet, but that's my next step. 2) I've got two 802.11b access points in my house, same SSID, one on channel 1, another on channel 6. The phone seems to stay associated with the access point that it first registered on, unless I do restart network on the phone. Even when the other access point is much closer with a much better signal. Both my Windows and Mac OS laptops switch between APs at will, depending on which is stronger -- seems like the phone should do the same. Anyone else noticed these issues? Also, I've got firmware v1.5.2 on my phone. Was trying to find a link to see if there's anything more recent. Anyone have a newer version or know where to get one? Thanks, Jim Meehan Oakland, CA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax detect/transfer problem solved
This took me forever to figure out. I could not get asterisk to detect inbound fax calls from my SIP provider. Voice calls would go through my dialplan fine. With a fax call, I'd see the call in the log, asterisk would answer it, and then it would mysteriously drop. I went through everything I could possibly think of. Turns out my SIP provider (race.com) has an E-Fax feature that detects and intercepts fax calls and stores them on their server (or e-mails them out to wherever you want). So after asterisk answers, race.com detects a fax, redirects the call to their fax receiver, and drops the call to my asterisk box. Trying to figure out now if there's any way to disable that feature with them. Their control panel portal doesn't seem to offer any way to disable it. Just thought I'd post my discovery here in case anyone else is tearing their hair out over this :) -j ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detect/transfer problem?
Trying to get my asterisk system to detect and transfer inbound fax calls.
System is supposed to detect the fax, and then ring my fax machine which is
connected to a Sipura device.
Here's my extensions.conf:
exten = 1000,1,Goto(s,1)
exten = 630446,1,Goto(s,3)
exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
exten = s,3,Answer
exten = s,4,Wait(3)
exten = s,5,Dial(${PHONES},30)
exten = s,6,Wait(2)
exten = s,7,Voicemail(u3001)
exten = s,8,Hangup
exten = fax,1,Dial(SIP/meehan-fax,30)
exten = fax,2,Congestion
exten = 2001,1,Dial(SIP/meehan-home,30)
exten = 2002,1,Dial(SIP/meehan-mobile,30)
exten = 2003,1,Dial(SIP/meehan-fax,30)
Detection seems to be working okay. If I call in with a voice call, my two
voice SIP phones ring as normal, etc. If I call in with a fax call, it seems
like Asterisk is detecting the fax correctly, but the fax never rings, and
the call is just dropped. Fax rings fine and is answered if I call ext. 2003
though. Here's the event log:
-- Executing AGI(SIP/1095-6694, callerid.agi|7753594712) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callerid.agi
-- AGI Script callerid.agi completed, returning 0
-- Executing SetCallerID(SIP/1095-6694, Haws Corporation 7753594712)
in new stack
-- Executing Answer(SIP/1095-6694, ) in new stack
-- Executing Wait(SIP/1095-6694, 3) in new stack
== Spawn extension (meehan, s, 4) exited non-zero on 'SIP/1095-6694'
Any ideas here?
Thanks,
Jim
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[Asterisk-Users] Fax detect/transfer problem?
---BeginMessage---
Trying to get my asterisk system to detect and transfer inbound fax calls.
System is supposed to detect the fax, and then ring my fax machine which is
connected to a Sipura device.
Here's my extensions.conf:
exten = 1000,1,Goto(s,1)
exten = 630446,1,Goto(s,3)
exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
exten = s,3,Answer
exten = s,4,Wait(3)
exten = s,5,Dial(${PHONES},30)
exten = s,6,Wait(2)
exten = s,7,Voicemail(u3001)
exten = s,8,Hangup
exten = fax,1,Dial(SIP/meehan-fax,30)
exten = fax,2,Congestion
exten = 2001,1,Dial(SIP/meehan-home,30)
exten = 2002,1,Dial(SIP/meehan-mobile,30)
exten = 2003,1,Dial(SIP/meehan-fax,30)
Detection seems to be working okay. If I call in with a voice call, my two
voice SIP phones ring as normal, etc. If I call in with a fax call, it seems
like Asterisk is detecting the fax correctly, but the fax never rings, and
the call is just dropped. Fax rings fine and is answered if I call ext. 2003
though. Here's the event log:
-- Executing AGI(SIP/1095-6694, callerid.agi|7753594712) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callerid.agi
-- AGI Script callerid.agi completed, returning 0
-- Executing SetCallerID(SIP/1095-6694, Haws Corporation 7753594712)
in new stack
-- Executing Answer(SIP/1095-6694, ) in new stack
-- Executing Wait(SIP/1095-6694, 3) in new stack
== Spawn extension (meehan, s, 4) exited non-zero on 'SIP/1095-6694'
Any ideas here?
Thanks,
Jim
---End Message---
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[Asterisk-Users] CallerID name lookup AGI script
Hi all,
My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:
1) If it's a toll free number (800|888|877|866), set the CallerID name to
TollFree Caller
2) Use curl to look up the number in Google phonebook
3) If a business listing, set the CallerID name to business name, as is.
4) If it's a residential listing, reverse the listing so it's last name first,
then set the CallerID name to that.
5) If there's no match in Google phonebook, look up the NPA/NXX on
www.areacodedownload.com and set the CallerID name to @ST RATECENTER where
ST is the two-letter state abbreviation, and RATECENTER is the name of
telco rate center in that state.
Thought some of you might find this AGI script useful, so I'm including it
below. It requires the Asterisk::AGI perl module.
There are other reverse phone lookup sources that are more complete than
Google's, but they are harder to screen scrape. Also, I probably could have
made this a little cleaner if I used the Google API rather than screen
scraping with curl/perl. Please feel free to take a shot at making any of
those modifications.
Here's a snippet from my extensions.conf where it gets called:
exten = s,1,AGI(callerid.agi|${CALLERIDNUM})
exten = s,2,SetCallerId,${googlename} ${CALLERIDNUM}
exten = s,3,Dial(${PHONES},30,r)
exten = s,4,Answer
exten = s,5,Wait(2)
exten = s,6,Voicemail(u3001)
exten = s,7,Hangup
And here's the script:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$number = $ARGV[0];
if ($number =~ m/(800|888|877|866)\d{7}/) {
$AGI-set_variable('googlename', \TollFree Caller\);
exit 0;
}
open(RESULTS, /usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.google.com/search
?q=phonebook:$number |);
while (RESULTS) {
if (m/Residential Phonebook/) {
$reverse = 1;
@fields = split(//);
}
if (m/Business Phonebook/) {
@fields = split(//);
}
if (m/did not match any/) {
@digits = split(//, $number);
$npa = $digits[0] . $digits[1] . $digits[2];
$nxx = $digits[3] . $digits[4] . $digits[5];
open(LOCATION, /usr/bin/curl -s -m 2 -A Mozilla/4.0 http://www.areacodedown
load.com/$npa/$nxx/ |);
while (LOCATION) {
if (m/State/) {
$line = LOCATION;
$line =~ m/\\#CACACA\\w* (\w\w)\/td/;
$name = [EMAIL PROTECTED];
}
if (m/Rate Center/) {
$line = LOCATION;
$line =~ m/\\#CACACA\((\w|\s)*)\/td/;
$name = $name . . $1;
}
}
$AGI-set_variable('googlename', \$name\);
exit 0;
}
}
@result = split(/-/, $fields[35]);
chop($result[0]);
if ($reverse) {
@words = split(/ /, $result[0]);
$last = pop(@words);
unshift(@words, $last,);
foreach $word (@words) {
$name = $name . $word . ;
}
}
if ($reverse == 0) {
$name = $result[0];
}
$AGI-set_variable('googlename', \$name\);
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