Re: [asterisk-users] Paetec SIP Trunk
We have customers that have migrated to our network from them due to their reliability issues. Most of them are in the US west and east. Jared Baxley jared.bax...@gmail.com wrote: Has anyone had experience using a SIP trunk provided by Paetec over MPLS? With or without FreePBX Regards, Jared Baxley -- _ -- 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] Hangup Cause and SIP Response Code
Kevin
I am using 1.8.x 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has been
reimplemented for Asterisk 11 in a way that doesn't affect performance as
much as the old method did).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Best non polycom SIP conference room phone
Thank you for your responses. No where did I say I hate polycom phones. I personally do not like their approach to sip as a company. Their audio quality is top notch but for me the rest leaves me wanting. Has anyone used the newer snom conference room phone? Bryant Zimmerman On Jan 8, 2012, at 10:59 AM, C F shma...@gmail.com wrote: I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality. However check out Panasonic. They make a sip conference phone. On 1/5/12, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com wrote: On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com wrote: I am looking for a really good SIP conference room phone for use with asterisk. I do not like Polycom at all. You have a really bad taste. There was an interesting flamewar one day in the Asterisk IRC channel over Polycom love/hate. We fall into the hate category here, and hope to never have to deal with them. If there was an SPA-series conference phone, we'd all rejoice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Microsoft Lync server and Asterisk access
Yes you can. Lync can not do registration and it is a trick to setup. Bryant On Apr 14, 2011, at 11:23 AM, Jim Dickenson dicken...@cfmc.com wrote: We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to connect to an Asterisk box? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- 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] Asterisk, Sent accountcode between 2 asterisk
On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten = _X.,3,Hangup
anyone know if it's possible to add the CDR Accountcode to this process
for get it on the second server B ?
i want the same accountcode on the 2 servers
thanks
Olivier
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Send the account code as a custom header variable encode it on A and read it on
B. You can send any variables you want using this method. I currently send
about 10 variables on switch transfers. If you need an example ping me back and
I will send one when I get in the office.
Bryant
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Re: [asterisk-users] Asterisk, Sent accountcode between 2 asterisk
On Mar 5, 2011, at 8:52 AM, brya...@zktech.com wrote:
On Mar 5, 2011, at 2:29 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten = _X.,3,Hangup
anyone know if it's possible to add the CDR Accountcode to this process
for get it on the second server B ?
i want the same accountcode on the 2 servers
thanks
Olivier
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Send the account code as a custom header variable encode it on A and read it
on B. You can send any variables you want using this method. I currently send
about 10 variables on switch transfers. If you need an example ping me back
and I will send one when I get in the office.
Just noticed you are using IAX I don't think my method works with IAX. That is
why I use SIP between systems. Someone correct me if there is a way to send
custom variables with IAX.
Bryant
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Re: [asterisk-users] res_fax
On Jan 20, 2011, at 8:53 PM, Steve Underwood On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve Steve thanks for your response. Do I need a copy of spandsp installed or is the res_fax_spandsp.so the complete package. If I need spandsp what version should I be using? The version I compiled and am using is now over a year old spandsp-0.0.5pre4. Where can I get the current stable version with a list of dependencies for compilation? Thanks Bryant -- _ -- 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] DIALSTATUS on CANCEL
If a call is hung up before an answer our h extension is not running in our
dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] DIALSTATUS on CANCEL
I am using the g option and it does not run the next statement or h extension
if the caller hangs up before an answers or time out event occurs during a
dial comand.
Bryant
On Dec 24, 2010, at 9:55 AM, Jim Dickenson dicken...@cfmc.com wrote:
If on the dial command you add option g, if the call is not answered, it will
fall through to the next statement which can be a hangup command and then it
will go to the h extension. If that does not then make the statement after
the dial command a goto h extension.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 24, 2010, at 6:03 AM, brya...@zktech.com wrote:
If a call is hung up before an answer our h extension is not running in
our dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] Wireless Desktop VoIP Phone?
I use grandstream with the linksys/cisco adapter. Bryant On Dec 17, 2010, at 3:04 PM, Michael Graves mgra...@mstvp.com wrote: On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? I beleive that snom supports the use of a wifi usb dongle in the 8x0 series phones. Also, Linksys/Cisco offered an 802.11g adapter that could be paired with their phones, making them wifi capable. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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
