Re: [asterisk-users] Nortel pbx & dtmf issues

>> I see from the archives that the issue of nortel handsets not >> sending dtmf tones to asterisk has been discussed a couple of times, >> but there is something I quite havent seen answered yet. >> >> Is this dtmf issue a problem with the nortel handsets or the PBX >> itself? If the handset were

Re: [asterisk-users] Using the PBX Directory from a Blackberry

On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote: > "Anyhow, on the Blackberry, when you hold down the Alt key and press the > alpha character, the device sends out the correct digit tone associated with > that character, like on a regular phone keypad." Any chance of answering the

[asterisk-users] Some IAX calls do not disconnect.

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients a

Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?

On Thu, 2009-07-02 at 17:42 -0400, John A. Sullivan III wrote: > On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote: > > > > > > On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III > > wrote: > > > > On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: > >

Re: [asterisk-users] Welcome Message

exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1}) exten => _X48600XXX,2,Hangup exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1}) exten => _X38600XXX,2,Hangup exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1}) exten => _X28600XXX,2,Hangup exten => _X18600XXX,1,Meet

Re: [asterisk-users] Dial

On Fri, Jul 3, 2009 at 12:02 AM, Steve Edwards wrote: > On Thu, 2 Jul 2009, Steve Totaro wrote: > >> AMI has historically been flaky under high load. >> >> Not sure about these days, probably better, but once upon a time >> pounding on the AMI would crash or just freeze a box, requiring a hard >> r

[asterisk-users] Converged mail box sizes

Just a thought as we explore the brave new world of converged voice and emails. Voice mail boxes typically hold a very small number of messages while email folders contain thousands. Do we need to rethink the traditionally small limits on voice mail boxes when storing in IMAP or are the messages

Re: [asterisk-users] Zimbra IMAP authentication - SOLVED

On Thu, 2009-07-02 at 20:59 -0400, John A. Sullivan III wrote: > Hello, everyone. No need to read this message. I'm posting for > documentation for other poor, ignorant slobs like me who are struggling > to pull together the many technologies to make converged networks > happen. Hopefully, this

Re: [asterisk-users] Dial

On Thu, 2 Jul 2009, Steve Totaro wrote: > AMI has historically been flaky under high load. > > Not sure about these days, probably better, but once upon a time > pounding on the AMI would crash or just freeze a box, requiring a hard > reboot. The machine would become totally unresponsive except

Re: [asterisk-users] Using the PBX Directory from a Blackberry

From: Darrick Hartman Sent: Thursday, July 02, 2009 8:06 PM On 07/02/2009 10:14 AM, Tzafrir Cohen wrote: > On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: >> Hi All, >> >> A couple of customers called complaining that folks were dialing into >> their PBX trying to use the Directory

Re: [asterisk-users] Dial

On Thu, Jul 2, 2009 at 8:51 PM, Steve Edwards wrote: > Un-top-posting... > >>> On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote: >>> I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my cal

Re: [asterisk-users] Using the PBX Directory from a Blackberry

It is a problem with Windows mobile phones as well, there is *NO* way to dial a number e.g. 800-CALL-ATT. On my Nokia S60 phone (E71) I can dial the number but it is not possible to dial letters when the call is connected. This affects everyone. When I call American Express it asks me to enter my

[asterisk-users] DAHDI

I finally decided it's been long enough using my ancient HP junker and I built a Atom 330 based machine to replace it. I've installed Centos 5, Dahdi and Asterisk 1.6.2. After a bit of struggles getting the 1.2 version files converted to 1.6 almost everything seems to be working. All my SIP li

Re: [asterisk-users] Using the PBX Directory from a Blackberry

On 07/02/2009 10:14 AM, Tzafrir Cohen wrote: > On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: >> Hi All, >> >> A couple of customers called complaining that folks were dialing into >> their PBX trying to use the Directory to locate users, from a >> Blackberry, and getting frustrated

[asterisk-users] Zimbra IMAP authentication - SOLVED

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email

Re: [asterisk-users] Dial

Un-top-posting... >> On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote: >> >>> I want to know the options that I have as an Asterisk user to make a >>> call and control its progress. I have been using the spool directory >>> to originate my calls using this format: >>> >>> Is there any other way to ori

Re: [asterisk-users] DUNDi Errors (ENCREJ)

On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote: > -> To resolve this i tried to remove all keys in all servers and once > again created and >distributed the loaded in each system with "keys init" command but > stilll i am >getting the same error > > > > can anybody help me out

Re: [asterisk-users] Dial

Thank you Steve. AMI is what I used before and now I am looking for something better if exists. On Thu, Jul 2, 2009 at 8:20 PM, Steve Edwards wrote: > On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote: > > > I want to know the options that I have as an Asterisk user to make a call > > and control its p

Re: [asterisk-users] Dial

On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote: > I want to know the options that I have as an Asterisk user to make a call > and control its progress. > I have been using the spool directory to originate my calls using this > format: > > Is there any other way to originate it and being able to cancel

[asterisk-users] Dial

Hello list, I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my calls using this format: "Channel: " "Callerid: " "MaxRetries:" "RetryTime: " "WaitTime: " "Context: " "Extension: " "Priority: "

Re: [asterisk-users] Rajkiran Reddy sent you a Friend Request on Yaari

Fail. Rajkiran Reddy wrote: > Rajkiran Reddy wants you to join Yaari! > > Is Rajkiran your friend? > > Yes, Rajkiran is my friend! > > > No, Rajkiran isn't my friend. >

Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?

On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote: > > > On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III > wrote: > > On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote: > > Hello, all. With the assistance of very helpful folks, our > b

Re: [asterisk-users] Using a mobile phone via USB as an extension

Carlos Ruiz Diaz a écrit : > Check chan_mobile. > [...] Or use GSM gateway > On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill wrote: > > >> I have had a search for this, but didn't come up with any results, so maybe >> I am >> using the wrong terms, sorry if this is an FAQ. >> >> For those who wan

Re: [asterisk-users] Using a mobile phone via USB as an extension

Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill wrote: > I have had a search for this, but didn't com

asterisk-users@lists.digium.com

Make sure this is in your xml config: ip.of.ntp.server Unicast On Thu, Jul 2, 2009 at 10:27 AM, mahboob zaman wrote: > > > -- Forwarded message -- > From: mahboob zaman > Date: Tue, Jun 30, 2009 at

[asterisk-users] Rajkiran Reddy sent you a Friend Request on Yaari

Rajkiran Reddy wants you to join Yaari! Is Rajkiran your friend? http://yaari.com/?controller=user&action=mailregister&friend=1&sign=YaariLBF849RQF972ZTA396ZMZ718";>Yes, Rajkiran is my friend! http://yaari.com/?controller=user&action=mailregister&friend=0&sign=YaariLBF849RQF972ZTA396ZMZ718";>No,

Re: [asterisk-users] Why Asterisk + Kamailio ?

Are there any good tutorials or overviews on a basic setup using a SIP router in conjunction with Asterisk? I would love to get a proof of concept up and working as I'm in the midst of a PBX re-architecture, and having the load-balancing/high availability features that a SIP frontend would provide

[asterisk-users] Using a mobile phone via USB as an extension

I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This pro

[asterisk-users] need help, service unavailable, registered but call does not get through

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701'

Re: [asterisk-users] AGI Transfer?

On Thu, 2 Jul 2009, J. G. wrote: > I've been trying to get an AGI "transfer" to work for several weeks now. > It isn't error-ing out, but it isn't working either. > > Does an AGI Transfer actually work? Does executing the transfer application in an AGI using the AGI exec command work? Yes. I e

Re: [asterisk-users] Why Asterisk + Kamailio ?

On Thu, 2 Jul 2009, jonas kellens wrote: > Why do I see many setups where there is an Asterisk server in > combination with a SIP-server like OpenSER or Kamailio ? > > Why is there an extra SIP server implemented at many VoIP-providers ?? OpenSER/Kamailio are supposed to be much better dealing wi

Re: [asterisk-users] * as VM for legacy PBX?

> Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The > VM box, itself, is beginning to show its age. Big-time. We're thinking > it > might be time to look for a replacement. I'd love to install Asterisk > with an FXO card or something, but I don't think it supports whate

[asterisk-users] Why Asterisk + Kamailio ?

Why do I see many setups where there is an Asterisk server in combination with a SIP-server like OpenSER or Kamailio ? Isn't Asterisk enough as SIP-server ?? It can communicate with many databases through ODBC, with many other software through an API (AGI), with other servers like OpenFire for jabb

Re: [asterisk-users] Extension status as XML for an Aastra 57i

So what I'm gathering is this I have to map each extension to a button, whether physical on a 560m or 536m or virtual using the soft buttons on the phone. What I was hoping for was something like the Directory app http://voip-pbx/aastra/directory.php that came with the phone's firmware that shows e

[asterisk-users] Grandstream 2010 and blinky lights

I am using 1.4, and have the above device, and it worked really well with monitoring 18 "hints" aka devices. Now, I've moved us to a hotdesking paradigm where the user is the "extension" not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF

Re: [asterisk-users] g729a compatibility

Hello! My Goodness! Thanks for the help! I did set up a SIP account for that device to allow G729, but it doesn't seem to be associating the device with that account. So that is where my confusion came from. I'll make the changes and see what happens. Regards, Elliot On Thu, Jul 2, 2009 at 7:

Re: [asterisk-users] g729a compatibility

Elliot Murdock wrote: > [Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) --- > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 : > 5060 (no NAT) > [Jul 2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as > basis request - 699864475636237-124654

Re: [asterisk-users] Echo and static on PRI with errors

On Thu, Jul 2, 2009 at 4:41 PM, Tzafrir Cohen wrote: > On Thu, Jul 02, 2009 at 04:15:13PM +0100, Tom O'Connor wrote: > > > I have tried all suggestions given. It just happens that none of them > have > > been much use. I'm very constrained by time on this project, less than > 11 > > days before

Re: [asterisk-users] g729a compatibility

Hello Everybody! Here are the full SIP logs! <--- SIP read from 216.48.184.50:5060 ---> INVITE sip:6587972772285...@82.80.231.238:5060;user=phone SIP/2.0 Call-ID: 699864475636237-1246542986-18105 From: ;tag=24794 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 216.48.184.5

Re: [asterisk-users] Authentication Issue Between Servers

No one ever responded to this inquiry but I figured out what the issue was. I thought I would respond with the solution just in case someone runs into the same issue in the future. Firstly, when setting up trunking between servers the "username =" field is not optional. :) Also, I had a lot

asterisk-users@lists.digium.com

-- Forwarded message -- From: mahboob zaman Date: Tue, Jun 30, 2009 at 8:42 AM Subject: cisco phone 7911 & To: asterisk-users@lists.digium.com Hellow, I have cisco 7911 and 7906 worked with asterisk server. But i can not set the time and date for these phones. can any one tell m

Re: [asterisk-users] Echo and static on PRI with errors

On Wed, Jul 1, 2009 at 6:58 PM, Steve Totaro wrote: > On Wed, Jul 1, 2009 at 6:35 AM, Tom O'Connor wrote: > > > > > > On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro < > stot...@first-notification.com> wrote: > >> > >> > >> On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor wrote: > >>> > >>> > >>> On W

Re: [asterisk-users] g729a compatibility

Elliot Murdock wrote: > Any help resolving this is would be great! Even though we've asked, you have not given us what we asked for. The most important thing to debug this problem is a complete console log, with 'core set verbose 10', 'core set debug 10' and 'sip set debug on', and ensure the the

Re: [asterisk-users] Using the PBX Directory from a Blackberry

On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: > Hi All, > > A couple of customers called complaining that folks were dialing into > their PBX trying to use the Directory to locate users, from a > Blackberry, and getting frustrated due to the incompatibility of > dialing alpha char

Re: [asterisk-users] g729a compatibility

Hello! Here is the 200 OK response from my server: Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 10646 10646 IN IP4 212.80.21.238 Owner Username: root Se

[asterisk-users] Using the PBX Directory from a Blackberry

Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of co

Re: [asterisk-users] Echo and static on PRI with errors

On Wednesday 01 July 2009 05:35:08 Tom O'Connor wrote: > On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro wrote: > > That is one option. The new line Digium cards are on par with Sangoma as > > far as IRQ issues. > > I don't really know what you mean about the new line Digium cards.. which > models

Re: [asterisk-users] g729a compatibility

Jeff LaCoursiere wrote: > Also - please post the output of the CLI command: > > core show translation That won't affect the SDP negotiations; if sip.conf allows g729 for that endpoint, Asterisk will negotiate it, regardless of whether there is a transcoding path available or not. Asterisk cannot

Re: [asterisk-users] g729a compatibility

Elliot Murdock wrote: > Which RFC specifies the corresponding number of the formats? Have you heard of Google? It's amazing in that it can find stuff for you like that faster than you can post to a mailing list and wait for a response :-) > Where in the Asterisk source code does it state the SDP

Re: [asterisk-users] g729a compatibility

On Thu, 2 Jul 2009, Jeff LaCoursiere wrote: On Thu, 2 Jul 2009, Elliot Murdock wrote: Hello Jeff, Yes, I use G729 all the time. Here is the SDP extrace from Wireshark. I'll get more data as it becomes available: Session Description Protocol Session Description Protocol Versio

Re: [asterisk-users] g729a compatibility

On Thu, 2 Jul 2009, Elliot Murdock wrote: Hello Jeff, Yes, I use G729 all the time. Here is the SDP extrace from Wireshark. I'll get more data as it becomes available: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (

Re: [asterisk-users] /var/lib/asterisk/sounds does not exist

On Wed, Jul 1, 2009 at 7:03 PM, bilal ghayyad wrote: > > Hi All; > > I download asterisk, compiled it and install it, but not finding the sounds > file (/var/lib/asterisk/sounds), what could be the reason and how I can have > it without repeating every thing? > > My asterisk version is: Asterisk

Re: [asterisk-users] g729a compatibility

Hello Jeff, Yes, I use G729 all the time. Here is the SDP extrace from Wireshark. I'll get more data as it becomes available: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): MG4000|2.0 49743 83164 IN IP4 216.48.184.

Re: [asterisk-users] /var/lib/asterisk/sounds does not exist

Just do your make menuselect and make install commands again, making sure you select sounds in the selection process. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Wednesday, July 01, 2009

Re: [asterisk-users] g729a compatibility

On Thu, 2 Jul 2009, Elliot Murdock wrote: Hello! Which RFC specifies the corresponding number of the formats? Where in the Asterisk source code does it state the SDP formats? Does Asterisk follow the formats of IANA? (http://www.iana.org/assignments/rtp-parameters) Thank you, Elliot Perha

Re: [asterisk-users] g729a compatibility

Hello! Which RFC specifies the corresponding number of the formats? Where in the Asterisk source code does it state the SDP formats? Does Asterisk follow the formats of IANA? (http://www.iana.org/assignments/rtp-parameters) Thank you, Elliot On Thu, Jul 2, 2009 at 3:44 PM, Elliot Murdock wrot

Re: [asterisk-users] Registrations problems to SIP-provider.

How about putting nat=yes for everything even if it isn't NAT, use qualify, and get rid of your firewall rules if you are registering. On Thu, Jul 2, 2009 at 8:18 AM, hh174 wrote: > Just by blocking the packets coming from 3stars. > Your asterisk, by receiving no responses think thet the host is n

Re: [asterisk-users] g729a compatibility

Elliot Murdock wrote: > However, if that is the case, why is Asterisk sending back PCMU > packets (instead of G729), which the device is not enabled for and > subsequently, fails the call? > > Could the mapping be disabled or not properly mapping to the G729 > driver in a certain versions of Aste

Re: [asterisk-users] g729a compatibility

Hello, Thank you clarifying that. However, if that is the case, why is Asterisk sending back PCMU packets (instead of G729), which the device is not enabled for and subsequently, fails the call? Could the mapping be disabled or not properly mapping to the G729 driver in a certain versions of Ast

[asterisk-users] AGI Transfer?

I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer ca

Re: [asterisk-users] g729a compatibility

Hello Kevin, Gotcha...however, the mime is G.729a with an extra period, so it doesn't get recognized. However, as I asked before, does Asterisk map any of the RTP/AVP profiles or does every format need to be defined in the SDP with a rtpmap attribute? You see, the incoming SDP supplies format 18

Re: [asterisk-users] g729a compatibility

Elliot Murdock wrote: > Hello! > > I noticed that the SIP packet contains this line: > > m=audio 6 RTP/AVP 18 98 96 97 101 13 > > However, there is no rtpmap that describes 18. Media format 18 > Apparently refers to G729, but there is no rtpmap in the SDP for it. > Since G729 is a registere

Re: [asterisk-users] g729a compatibility

Philipp Kempgen wrote: > http://tools.ietf.org/html/rfc3555#section-3 > ---cut--- >Note that the payload format (encoding) names defined in the RTP >Profile are commonly shown in upper case. MIME subtypes are commonly >shown in lower case. These names are case-insensitive in both >

Re: [asterisk-users] Registrations problems to SIP-provider.

Just by blocking the packets coming from 3stars. Your asterisk, by receiving no responses think thet the host is not reachable. They are mayny good routers/firewall permitting to avoid these kind of problems. Olivier jonas kellens a écrit : Actually it was my Firewall (Endian). By reb

Re: [asterisk-users] Registrations problems to SIP-provider.

Actually it was my Firewall (Endian). By rebooting my firewall, all problems were solved and till this moment every communication succeeds. I do expect them back... I don't want to hijack this Asterisk-mailinglist, but I think that firewall-issues also are related to Asterisk-support. So my quest

Re: [asterisk-users] Force Authentication

Dear Sir, I have a Patton device registered on an OpenSIPs server...The openSips Server send the call to asterisk serverWhen using some end user device like patton the asterisk server does not send back an Authentication required packet to INVITE packet received from OS as follow: <--- SIP re

Re: [asterisk-users] UK Vodafone femtocells now available

or maybe i misread :) 2009/7/1 Mike Dent > 2009/7/1 Geraint Lee : > > agreed. > > > > extended o2 coverage would be very useful, especially for Wales! > > > > I like the idea, i don't like the idea of paying, if they want mobile > > traffic it should be possible to buy your own hardware controll

Re: [asterisk-users] Nortel pbx & dtmf issues

> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Ernest Byaruhanga > Sent: Thursday, July 02, 2009 4:10 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Nortel pbx & dtmf issues >

Re: [asterisk-users] g729a compatibility

Hello Phillip! Thank you. However, do you what Asterisk does if there is no rtpmap that describes the format profile in the media description? The SIP packets states: audio 25184 RTP/AVP 18 98 96 97 101 13 However, there is no rtpmap for format 18. Asterisk does not seem to be associating 18

[asterisk-users] Ext1: Channel X parameter on PRI

Hello, I use asterisk as a PRI gateway between the PRI line from the telco (span 1) and the customers PBX (span 2) . I did a "pri debug" because I had some problems with incoming routings. >From the telco I get: 1 < Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0

Re: [asterisk-users] g729a compatibility

Hello! Oops, a typo: Hello! I noticed that the SIP packet contains this line: m=audio 6 RTP/AVP 18 98 96 97 101 13 However, there is no rtpmap that describes 18. Media format 18 Apparently refers to G729, but there is no rtpmap in the SDP for it. Since G729 is a registered and known forma

Re: [asterisk-users] g729a compatibility

Hello! I noticed that the SIP packet contains this line: m=audio 6 RTP/AVP 18 98 96 97 101 13 However, there is no rtpmap that describes 18. Media format 18 Apparently refers to G729, but there is no rtpmap in the SDP for it. Since G729 is a registered and known format is there any way for

Re: [asterisk-users] g729a compatibility

Elliot Murdock schrieb: > Thank you for that piece of information. Which RFC does it state that > the audio name is "G729"? http://tools.ietf.org/html/rfc3555#section-4.1.9 > On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Fleming wrote: >> Elliot Murdock wrote: >>> I have a sip device that is sending

[asterisk-users] Nortel pbx & dtmf issues

folk, I see from the archives that the issue of nortel handsets not sending dtmf tones to asterisk has been discussed a couple of times, but there is something I quite havent seen answered yet. Is this dtmf issue a problem with the nortel handsets or the PBX itself? If the handset were changed to