Re: [asterisk-users] PJSIP Dial via IP fails
CDR wrote:
Dear friends
This is my simple dialplan
[demopjsip]
exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten = _X.,n,Hangup()
Currently the default outbound endpoint is only used for messaging. This
may change in the future but as of right now it is not used for
sessions. You will therefore have to explicitly specify it:
Dial(PJSIP/default_outbound_endpoint/sip:${EXTEN}@10.10.10.2)
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
The following is a basic syntax of what you need to do before the Dial function is launched: Set(CALLERID(num)=1234567890) More details is here: http://www.voip-info.org/wiki/view/Asterisk+func+callerid Regards HASSAN On Thu, Jun 26, 2014 at 8:10 PM, Positively Optimistic positivelyoptimis...@gmail.com wrote: We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
It depends on your carrier.With some carriers, such as Verizon SIP, you do this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how to bill the call, it will be rejected. With Level 3 SIP, you can use From: or PAID but if the number you present to them is not on your account, they will bill the call against the “failsafe” number on your account. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic Sent: Thursday, June 26, 2014 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
Thanks Hassan, That is precisely what we are doing... if we Set(CALLERID(num)=8002211212) the call fails.. however, if we Set(CALLERID(num)=256963800) it is successful. The call will complete if we use any valid non-toll free number in the LERG. If we use a toll free, it fails. On Thu, Jun 26, 2014 at 9:15 AM, Nyamul Hassan nya...@gmail.com wrote: The following is a basic syntax of what you need to do before the Dial function is launched: Set(CALLERID(num)=1234567890) More details is here: http://www.voip-info.org/wiki/view/Asterisk+func+callerid Regards HASSAN On Thu, Jun 26, 2014 at 8:10 PM, Positively Optimistic positivelyoptimis...@gmail.com wrote: We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
You need to talk to your carrier. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic Sent: Thursday, June 26, 2014 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info Thanks Hassan, That is precisely what we are doing... if we Set(CALLERID(num)=8002211212) the call fails.. however, if we Set(CALLERID(num)=256963800) it is successful. The call will complete if we use any valid non-toll free number in the LERG. If we use a toll free, it fails. On Thu, Jun 26, 2014 at 9:15 AM, Nyamul Hassan nya...@gmail.commailto:nya...@gmail.com wrote: The following is a basic syntax of what you need to do before the Dial function is launched: Set(CALLERID(num)=1234567890) More details is here: http://www.voip-info.org/wiki/view/Asterisk+func+callerid Regards HASSAN On Thu, Jun 26, 2014 at 8:10 PM, Positively Optimistic positivelyoptimis...@gmail.commailto:positivelyoptimis...@gmail.com wrote: We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume the mediagateway will convert the headers to something that PRI can understand. On Thu, Jun 26, 2014 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote: It depends on your carrier.With some carriers, such as Verizon SIP, you do this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how to bill the call, it will be rejected. With Level 3 SIP, you can use From: or PAID but if the number you present to them is not on your account, they will bill the call against the “failsafe” number on your account. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Positively Optimistic *Sent:* Thursday, June 26, 2014 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing recorded file storage directory.
Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP question
In a PJSIP endpoint, how do I set all no-named settings so they get inherited from another place and I don't need to mention them again and again for all my endpoints? In regular sip you could specify those options and they remained valid if not redefined by a peer. A case would be the codecs allowed. I tried to include those global options in a section called [global] disallow=all allow=ulaw but the endpoints do not have knowledge of any such global options. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP question
You can use templates. Templates are defines by putting a “(!)” next to the context name. [template-name](!) disallow=all allow=ulaw Then define the template next to the endpoint in parenthesis. [endpoint-name](template-name) Chad On Jun 26, 2014, at 12:30 PM, CDR vene...@gmail.com wrote: In a PJSIP endpoint, how do I set all no-named settings so they get inherited from another place and I don't need to mention them again and again for all my endpoints? In regular sip you could specify those options and they remained valid if not redefined by a peer. A case would be the codecs allowed. I tried to include those global options in a section called [global] disallow=all allow=ulaw but the endpoints do not have knowledge of any such global options. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP Include not working
I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is interested to
pay the bill
exten = s,1,AGI(your-agi-program.pl)
exten = s,2,others...
Regards..
rv
2014-06-26 11:19 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged
otherwise not.
Thanks
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
--
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Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Thanks Rafeal. This is what I needed.
But first line i.e.
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
is very complicated.
I have very simple plan which is as below.
[context-demo]
exten=,1,AGI ( pythonscript.py )
exten=,1,Dial(SIP/)
that all.
Now can you please explain me in simpler form.
I am sorry. I am a newbie.
On Thu, Jun 26, 2014 at 11:12 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is interested to
pay the bill
exten = s,1,AGI(your-agi-program.pl)
exten = s,2,others...
Regards..
rv
2014-06-26 11:19 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged
otherwise not.
Thanks
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life in
the midst of these materialistic turbulences.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Executing an AGI python script in Asterisk after call is bridged.
Ok.
in this link you will find some easy macro
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Explanation
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINATION},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
Dial comand
to number DESTINATION
with timout of 20 seconds
r=ring sound
g= when hungs, continue with the dialplan (keep alive de application)
M=execute macro acceptcall passing the value SESSIONID
acceptcall=name of the macro
MAXCALLTIME is a value in second that the call is allowed to (prepaid)
So maybe you have to think somethin like this
exten=,1,Dial(SIP/,,rgM(mymacro))
and at the end of your dialplan
[macro-mymacro]
exten=s,1,AGI ( pythonscript.py )
It's not easy...
Good Luck.
2014-06-26 13:57 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Thanks Rafeal. This is what I needed.
But first line i.e.
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
is very complicated.
I have very simple plan which is as below.
[context-demo]
exten=,1,AGI ( pythonscript.py )
exten=,1,Dial(SIP/)
that all.
Now can you please explain me in simpler form.
I am sorry. I am a newbie.
On Thu, Jun 26, 2014 at 11:12 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is interested
to pay the bill
exten = s,1,AGI(your-agi-program.pl)
exten = s,2,others...
Regards..
rv
2014-06-26 11:19 GMT-04:00 Anurag Rana anuragrana31...@gmail.com:
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged
otherwise not.
Thanks
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life
in the midst of these materialistic turbulences.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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[asterisk-users] Originate with Caller ID Name
I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single value. In the Invite, the following line is essentially identical with the exception of the screen=no. Freeswitch caller ID name is successfully passing through but it passes screen=yes. Remote-Party-ID: Jane Done sip:8005551...@xxx.xxx.xxx.xxx;party=calling;privacy=off;screen=no For the AMI Originate, I have been passing variables in an attempt to modify the CALLERID(name-pres). My understanding is that a variable of CALLERID(name-pres)=allowed_passed_screen should result in the RPID screen setting being yes. I have tried many different values for this variable, but the RPID line is always screen=no. What am I missing to force the screen=yes to be passed as part of the Remote-Party-ID? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info
There seem to be a number of places number presentation could go wrong. Since the PRI 'used to' show toll-free numbers correctly, you need to look at the gateway. Can you debug the ISDN message on the gateway? See how the toll-free is being sent to the carrier. Since you are looking to manipulate the ISDN message via SIP, it all comes down to how the gateway handles the desired functions. Sincerely, Brian LaVallee On 6/26/14, 11:24 PM, Positively Optimistic wrote: We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume the mediagateway will convert the headers to something that PRI can understand. On Thu, Jun 26, 2014 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote: It depends on your carrier.With some carriers, such as Verizon SIP, you do this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how to bill the call, it will be rejected. With Level 3 SIP, you can use From: or PAID but if the number you present to them is not on your account, they will bill the call against the “failsafe” number on your account. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Positively Optimistic *Sent:* Thursday, June 26, 2014 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] CLID Presentation Billing Number | Diversion vs. Remote-Party-ID vs. P-Asserted-Id vs. From vs. P-Charge-info We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header for presentation? We can't possibly be the only people in the world that has faced this challenge. Searching the internet has provided limited results. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
