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davies147 at gmail.com
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 Posted: Fri Apr 25, 2008 8:08 am Post subject: [asterisk-users] No CallerID Transfer Problem |
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2008/4/24 Ken Williams <ken at intermountainelectronics.com>:
| Quote: |
Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with others,
but for some reason when they would transfer an outside call anywhere it
would pause for a few seconds and hang up the line.
Well, after spending most of the day on it, it turns out it's because they
don't have callerID on the PSTN lines coming in through zaptel. My first
thought was, set "usecallerid=no" and all would be well, but this didn't do
any good. After playing a bit longer I just set the following:
exten => 900,2,set(CALLERID(num)="606-555-1212")
exten => 900,3,set(CALLERID(name)="Outside Call")
exten => 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
Now all works well.
So is there another option somewhere to keep asterisk from killing a
transfer without callerid? This happened on both 1.4.17 & 1.4.18.1.
Thanks,
Ken
|
Can I guess that they are using snom phones with firmware 7.1.30? I
encountered exactly that bug here, but only if I enabled "sendrpid" in
the sip.conf of the asterisk system. Downgrading to a more-stable
6.5.x snom firmware, or disabling "sendrpid" for all of the snom
devices fixed this in our case. (Roll on the next snom firmware
release!)
If not, then can I suggest that you provide more detail of equipment
involved - PCI cards, handsets etc etc?
Hope that helps,
Steve |
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eric at fnords.org
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| Posted: Fri Apr 25, 2008 8:23 am Post subject: [asterisk-users] No CallerID Transfer Problem |
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Try removing the quotes from the Caller*ID info.
Steve Davies wrote:
| Quote: | 2008/4/24 Ken Williams <ken at intermountainelectronics.com>:
| Quote: | Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with others,
but for some reason when they would transfer an outside call anywhere it
would pause for a few seconds and hang up the line.
Well, after spending most of the day on it, it turns out it's because they
don't have callerID on the PSTN lines coming in through zaptel. My first
thought was, set "usecallerid=no" and all would be well, but this didn't do
any good. After playing a bit longer I just set the following:
exten => 900,2,set(CALLERID(num)="606-555-1212")
exten => 900,3,set(CALLERID(name)="Outside Call")
exten => 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
Now all works well.
So is there another option somewhere to keep asterisk from killing a
transfer without callerid? This happened on both 1.4.17 & 1.4.18.1.
Thanks,
Ken
|
Can I guess that they are using snom phones with firmware 7.1.30? I
encountered exactly that bug here, but only if I enabled "sendrpid" in
the sip.conf of the asterisk system. Downgrading to a more-stable
6.5.x snom firmware, or disabling "sendrpid" for all of the snom
devices fixed this in our case. (Roll on the next snom firmware
release!)
If not, then can I suggest that you provide more detail of equipment
involved - PCI cards, handsets etc etc?
Hope that helps,
Steve
_______________________________________________
-- 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
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--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
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ken at intermountainel...
Guest
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| Posted: Fri Apr 25, 2008 8:39 am Post subject: [asterisk-users] No CallerID Transfer Problem |
|
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Actually, the code below works perfectly to fix the transfer disconnect
problem. I was asking of other, better ways, aside from manually
defining on all incoming calls a dummy CID.
To answer Steve's question, using a single TDM400 card for the incoming
PSTN (it's one line, a remote office that most of their communication is
done over IAX back to our main location). The three handsets are
Grandstream GXP-2000 (let the flaming begin, we currently have about 40
GXP-2000's in production and yes, we've had strange issues, but they're
working quite well now).
Anyway, it's really not a huge deal, but I had work arounds. I'd prefer
the 'usecallerid=no' type route instead of making a fix in the dialplan,
that's all I was looking for.
Ken
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
Wieling
Sent: Friday, April 25, 2008 7:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No CallerID Transfer Problem
Try removing the quotes from the Caller*ID info.
Steve Davies wrote:
| Quote: | 2008/4/24 Ken Williams <ken at intermountainelectronics.com>:
| Quote: | Came upon a problem today that I thought I'd see if it's by design,
if I'm missing an option somewhere, or if my fix is the way to fix
|
|
it.
| Quote: | | Quote: |
We setup a remote location with a server, same as we've done with
others, but for some reason when they would transfer an outside call
anywhere it would pause for a few seconds and hang up the line.
Well, after spending most of the day on it, it turns out it's because
|
|
| Quote: | | Quote: | they don't have callerID on the PSTN lines coming in through zaptel.
|
|
| Quote: | | Quote: | My first thought was, set "usecallerid=no" and all would be well, but
|
|
| Quote: | | Quote: | this didn't do any good. After playing a bit longer I just set the
|
|
following:
| Quote: | | Quote: |
exten => 900,2,set(CALLERID(num)="606-555-1212")
exten => 900,3,set(CALLERID(name)="Outside Call")
exten =>
900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
Now all works well.
So is there another option somewhere to keep asterisk from killing a
transfer without callerid? This happened on both 1.4.17 & 1.4.18.1.
Thanks,
Ken
|
Can I guess that they are using snom phones with firmware 7.1.30? I
encountered exactly that bug here, but only if I enabled "sendrpid" in
|
| Quote: | the sip.conf of the asterisk system. Downgrading to a more-stable
6.5.x snom firmware, or disabling "sendrpid" for all of the snom
devices fixed this in our case. (Roll on the next snom firmware
release!)
If not, then can I suggest that you provide more detail of equipment
involved - PCI cards, handsets etc etc?
Hope that helps,
Steve
_______________________________________________
-- 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
|
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
_______________________________________________
-- 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 |
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