Re: [asterisk-users] Asterisk inside network. What phone works well?

On Thu, Oct 13, 2016 at 12:06 PM, wrote: > > I have Asterisk running well inside our network. I did some > > experiments exposing it to internet but had some issues: > > 1. NAT issues (voice one way, etc). From what I understand double- > > NAT users will always

Re: [asterisk-users] small pbx for the office [it was: small homebrew pbx]

On Wed, Jun 17, 2015 at 9:07 AM, lu...@sulweb.org wrote: Lukasz Sokol wrote: but have you considered a web-managed config-builder such as FreePBX? Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general

Re: [asterisk-users] Strange Polycom Issue

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote: Welcome to our hell. We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally got Polycom to issue a hotfix firmware version. I'll be happy to share it with you offlist, just email me. Officially

Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)

On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol el.es...@gmail.com wrote: As the handsets have no LCD's to show the dialled number, I want to give the workforce the ability to dial OUT using the softphone, (as in, copy/paste the number from the CRM software into softphone then *immediately*

Re: [asterisk-users] Asterisk LTS segment faults

On Wed, Oct 8, 2014 at 9:35 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, Does anyone know how frequent segment faults occur in the current LTS release (version 11) and in the future LTS release (version 13)? We are currently using 1.6, which frequently throws unexplained segment

[asterisk-users] Certified Asterisk 11.6 Menuselect

Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks, Ryan --

Re: [asterisk-users] DUNDi with SIP Mapping

On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: You are a bit outside of what I have done, but this looks like it might be what you want to do with SIP: http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP I had looked at that guide

Re: [asterisk-users] Live Recording on the Storage Server?

On Thu, Apr 17, 2014 at 11:52 AM, Bryant Zimmerman brya...@zktech.comwrote: A simple way that we use to do the move is to create a cron job that looks for a .move file. It has the same name as the recorded file. asterisk writes the .move file which is just a text file with some stats in it.

[asterisk-users] DUNDi with SIP Mapping

From the reading and testing I have done it doesn't look like SIP supports a username and password in the Dial string. I currently have the following mapping. priv = dundi-extens,0,SIP, dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial On the sending side I see NOTICE[31598]

Re: [asterisk-users] DUNDi with SIP Mapping

On Wed, Apr 16, 2014 at 9:06 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: I am using DUNDi with SIP to do some least cost routing amongst my various locations. My mapping is close to what you have: priv = dundi-extens,0,SIP,trunk_name/number_to_dial Where trunk_name is replaced

Re: [asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

On Thu, Jan 2, 2014 at 11:13 AM, Eric Wieling ewiel...@nyigc.com wrote: Which firmware version? 4.1.x is only for use with MS Link server. A symptom of running 4.1.x firmware with a non-MS server is the phone will not show buddies. I'm running 4.1.0 on a Polycom IP 335 and IP 550 and

Re: [asterisk-users] Maximum number of users

On Wed, Dec 18, 2013 at 9:45 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Can someone advise me what is the maximum number of users (IP Phones) that can be supported by asterisk 1.8 or later? Regards Bilal The number of devices and concurrent calls is dependent on many factors.

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On Sun, Dec 15, 2013 at 5:07 AM, jg webaccou...@jgoettgens.de wrote: I think the order or elements is relevant: [100] disallow=all allow=ulaw allow=g722 or [100] allow=!all,ulaw,g722 should work. jg If I choose that order and the phone supports both ulaw and g722 only ulaw will be

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On Sun, Dec 15, 2013 at 7:20 AM, jg webaccou...@jgoettgens.de wrote: I see, you do want something like picking g722 provided there is no transcoding. Because Asterisk is a B2BUA it can transcode, so it would choose g722 where the other party is doing g711. For known parties, maybe one could

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On Sun, Dec 15, 2013 at 9:32 AM, jg webaccou...@jgoettgens.de wrote: Is it possible to let the Sangoma card work only on the most demanding codecs? This requires some analysis to estimate the benefits. Another question is whether the user phones are provisioned or not. If provisioned, then

[asterisk-users] Why doesn't Asterisk try to prevent transcoding

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 - ulaw to ulaw is chosen 100 dials 101 - g722 to ulaw is

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner rswago...@gmail.com wrote: Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw

Re: [asterisk-users] Asterisk on Windows

On Wed, Dec 4, 2013 at 10:19 AM, CDR vene...@gmail.com wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep

Re: [asterisk-users] Trouble with upgrading - RBS T1

I have a system with two Sangoma A104D cards running Asterisk 1.8.11-cert10, Dahdi 2.5.0.1, LibPRI 1.4.12, and Wanpipe 3.5.23. The PRI spans are configured with esf,b8zs. Everything has been working great, which is why I haven't updated it further. You might try an older Dahdi version just to see.

Re: [asterisk-users] SIP Presence across two servers

I haven't tried it, but the res_corosync module states it will sync device state across servers. https://wiki.asterisk.org/wiki/display/AST/Corosync On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.com wrote: Aligning presence over multiple servers is not simple and require

Re: [asterisk-users] Suggestion of Server Specifications for Asterisk

On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote: Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? You want some

Re: [asterisk-users] T.38 gateway patch against Asterisk 1.8.11.0

On Fri, Mar 30, 2012 at 1:16 PM, Bryant Zimmerman brya...@zktech.comwrote: What does this patch fix? Why is it not in Jarr? Thanks Bryant It looks like the patch is a backport of the t.38 gateway functionality in Asterisk 1.10. Ryan --

Re: [asterisk-users] Stuck DAHDI Lines

2012/2/9 Antonio Modesto mode...@isimples.com.br ** Hi, Sometimes some of my dahdi channels become stuck, It is very strange, here is the output of the core show channels command: pabx*CLI core show channels Channel Location State Application(Data)

Re: [asterisk-users] Hang up phone after declined attended transfer

On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an

Re: [asterisk-users] Hang up phone after declined attended transfer

On Tue, Jan 10, 2012 at 1:57 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6

Re: [asterisk-users] Best non polycom SIP conference room phone

On Sun, Jan 8, 2012 at 12:03 PM, brya...@zktech.com wrote: 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

Re: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller

On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.com wrote: Doug: for what it's worth I am having the exact same nightmare. Not sure exactly when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am running 1.8.9rc1). I also have Polycom (335, 550, 650) and

Re: [asterisk-users] Anyone have a reliable T.38 Solution

On Fri, Jan 6, 2012 at 10:01 AM, Benny Amorsen benny+use...@amorsen.dkwrote: David Backeberg dbackeb...@gmail.com writes: Thanks for clearing that up. I was getting all excited that I could flash the PAP2T; I've always used regular voice tones over SIP with the PAP2Ts. SPA-2102 supports

Re: [asterisk-users] 1.6 and 1.8

On Thu, Dec 29, 2011 at 12:05 AM, Bruce B bruceb...@gmail.com wrote: I have been running 1.8.7 with a few fixes back ported from the 1.8.8 release candidate for the last 2.5 months. The system processes around 4,000 calls per day over PRIs for 250 Polycom phones. Previously I was running

Re: [asterisk-users] 1.6 and 1.8

On Thu, Dec 29, 2011 at 12:18 PM, Bruce B bruceb...@gmail.com wrote: Log are being filled with g729 transcoding error in 1.8.7x now :-( I don't dare to test 1.8.8x as it might have something else broken. Unfortunately I can no longer trust the release candidates. Thanks for the input. What

Re: [asterisk-users] 1.6 and 1.8

On Wed, Dec 28, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote: I understand the end of life issue. What I fail to understand is that if 1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8 have so many bugs (just what I read here, not from my actual experience)?

Re: [asterisk-users] Codec warnings after upgrade to 1.8

On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote: I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)? WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel

Re: [asterisk-users] CDR mysql with asterisk 1.4

On Mon, Nov 21, 2011 at 7:13 AM, salaheddine elharit salah.elharit...@gmail.com wrote: hello list i have asterisk 1.4 installed and i want to use CDR mysql  during the installation i didn’t check the cdr mysql with  make menuselect my question : i want to check this option now  after the

Re: [asterisk-users] Polycom Attended Transfer

On Wed, Nov 16, 2011 at 1:49 PM, eherr email.eherr9...@gmail.com wrote: When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller’s callerID during a blind transfer? Thanks, --E

Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburg l...@solvent-llc.com wrote: Interesting.  I just signed up with Gafachi (haven't even tested the service yet) but I planned to make use of their T38 support since they are listed at voip-info as being one of the ITSP's that _do_ support T38.  Have

Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

On Sat, Oct 8, 2011 at 3:51 PM, James Sharp ja...@fivecats.org wrote: On 10/08/2011 02:38 PM, Ryan Wagoner wrote: I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode

Re: [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?

On Wed, Jun 29, 2011 at 8:34 AM, Olivier oza_4...@yahoo.fr wrote: 2011/6/29 Ruben Rögels ruben.roeg...@jumping-frog.org Personally I would use HTTP too. Simple reason: You are much more flexible with it and a in most scnearios you have a webserver running anyway. I build some PHP-Script

Re: [asterisk-users] VMX Locator

On Thu, Jun 23, 2011 at 7:45 AM, Darrin Henshaw darrin.aster...@gmail.com wrote: Hello All, I've been doing some looking into VMX Locator(part of FreePBX from what I see). One of my sales guys came from a company that was running FreePBX and we are running straight asterisk installed using

Re: [asterisk-users] Connected Line ID

On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

On Sun, May 29, 2011 at 3:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: And f/w POS3-07-4-00 That is strange that Asterisk is not sending anything back in response to the register. Have you looked at the Asterisk console or logs to see why it is rejecting the register. You might have

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

On Mon, May 30, 2011 at 2:45 PM, Ian S. Worthington ianworthing...@usa.net wrote: Console is showing the following. Looks like it doesn't like the format of the REGISTER message??? --- SIP read from UDP:192.168.1.114:5060 --- REGISTER sip:192.168.1.41 SIP/2.0 Via: SIP/2.0/UDP

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

On Mon, May 30, 2011 at 5:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: Many thanks for that. I tried pedantic=no (adding it directly to the [702] section in sip_additional.conf: I'm using the freepbx frontend and it doesn't seem to have a way to enter that through the gui), but it

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

On Sat, May 28, 2011 at 4:08 PM, Ian S. Worthington ianworthing...@usa.net wrote: I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip

Re: [asterisk-users] Cisco registration problem with 1.8.3.3

On Sat, May 28, 2011 at 5:18 PM, Ian S. Worthington ianworthing...@usa.net wrote: I too had heard that 1833 did NOT have the 184 problem, which makes me suspicious that it's not that. I don't think its a NAT problem.  Neither a sip trace not tcpdump show any response at all to the incoming

Re: [asterisk-users] how to know how many calls are on hold

On Tue, May 17, 2011 at 10:16 AM, virendra bhati virbh...@gmail.com wrote: hi list, please help me how to know how many calls are on hold. If they are SIP channels you can use: sip show inuse The last column are calls on hold. Ryan --

Re: [asterisk-users] Polycom IP335 3.3.1 Call Waiting

On Thu, May 19, 2011 at 1:24 PM, Ryan Wagoner rswago...@gmail.com wrote: I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller

[asterisk-users] Polycom IP335 3.3.1 Call Waiting

I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with

Re: [asterisk-users] OPTIONS Keep alive - Reply: 481 No subscription

On Fri, May 13, 2011 at 2:58 PM, Skyler skchopper...@gmail.com wrote: Hi all,  Anyone know how to make asterisk properly reply to  options keep-alive? Or just force a 200 OK somehow?  I recently took over a server and they have ~80 pap2 devices that send nat keep-alive and * always

Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote: On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly

Re: [asterisk-users] Asterisk, SIP Firewalls

On Wed, Apr 27, 2011 at 1:16 PM, Myles Wakeham my...@techsol.org wrote: It kinda scares me though.  I know that SIP is an attractive attack-vector, and that there are scripts out there that target SIP devices.  I know I could run Fail2Ban on the server, which is fine (we're doing that anyway

Re: [asterisk-users] Warm Transfer in Asterisk

On Sat, Apr 23, 2011 at 8:56 AM, Jeff Johnson jjohn...@neturallyspeaking.com wrote: Is there a way do what is sometimes called a 3rd party transfer in Asterisk.  That is; Call A comes in and is answered B.  B then places A on hold and calls C.  After C answers, BC chat for a moment, then B

Re: [asterisk-users] 1.8.4-rc2: ReceiveFAX fails

On Fri, Apr 15, 2011 at 7:00 PM, sean darcy seandar...@gmail.com wrote: Using spandsp-0.0.6-pre18, the Jan 22 release. You might try using spandsp-0.0.6-pre17. That version works great for me with 1.8.4-rc2. When I tried pre18 it failed to receive any faxes. Ryan --

Re: [asterisk-users] Asterisk PRI back-to-back connect

On Tue, Mar 22, 2011 at 12:53 PM, satish patel satish...@hotmail.com wrote: Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before

Re: [asterisk-users] Need to buy the Digium card, to confirm

On Sat, Feb 26, 2011 at 5:33 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; My server and its slots written in it the following so I need to know which card to order it (I need a card supporting 2 E1s): PCIE_G2_X4 PCIE_G2_X8 Actually I do not know what is meaning by G2. OK I

Re: [asterisk-users] Paging with Polycom 3.3.x

On Thu, Feb 24, 2011 at 1:41 PM, Mike l...@net-wall.com wrote: Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried).  That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I

Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote: On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all.  I've finally made the jump from 1.4 to 1.8.  I've installed everything (I think), my

Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

On Sun, Feb 20, 2011 at 9:44 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote: On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all.  I've finally made

Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?

On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all.  I've finally made the jump from 1.4 to 1.8.  I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands.  I've

Re: [asterisk-users] Polycom IP335

On Thu, Feb 17, 2011 at 12:02 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: No parameters were rejected. Maybe my perception of backlight off is incorrect. When it is off I expect it so be similar to a Cisco 7961. So no light whatsoever and very hard to read in dim light. Yet in the

Re: [asterisk-users] Polycom IP335

On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote: I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it’s

Re: [asterisk-users] Polycom IP335

On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote: I have it on the 430s. I think it’s a firmware issue but I am having trouble replicating it on the 430 I could have sworn I had it on one phone before I rebooted it but memory might be influenced by hopes. What

Re: [asterisk-users] Polycom IP335

On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: I share your pain. I have an IP335 and IP670 here. Have not configured the IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670 running the latest bootrom (4.3.0) and firmware (3.3.1).

Re: [asterisk-users] Polycom IP335

On Wed, Feb 16, 2011 at 8:38 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 02/17/2011 12:10 AM, Ryan Wagoner wrote:   up     up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3     /up.backlight   /up Here's what I have: up up.idleTimeout=10

Re: [asterisk-users] Moving asterisk from one network to another.

On Fri, Dec 24, 2010 at 7:40 AM, Jim Dickenson dicken...@cfmc.com wrote: If you set bindaddr in any conf file you will need to change the IP address there. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ You will also need to change externip and localnet if those are

Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am using pfSense to do firewall and NAT on an Asterisk server. I have ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP 192.168.5.5. However, when a user from outside using Linksys

Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

On Sat, Dec 11, 2010 at 11:45 AM, Bruce B bruceb...@gmail.com wrote: Thanks for the feedback Ryan. Siproxd is not installed. I think Siproxd like you said just does the reverse meaning if phones are part of pfSense subnet then it connects to outside world. But in my case they are coming into

Re: [asterisk-users] Why does sip show peers show my router/gateway address as the client IP address?

On Sat, Dec 11, 2010 at 1:04 PM, Bruce B bruceb...@gmail.com wrote: Thanks for the confirmation. Do you have both LAN and WAN as outbound AON like this: WAN any * * * * * YES LAN  any * * * * * YES ??? I am stumped as to why pfSense behaves like this in this instance. Thanks again. You

Re: [asterisk-users] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities

On Wed, Aug 4, 2010 at 10:44 AM, Wouter Schoot wou...@schoot.org wrote: Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question. Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support

Re: [asterisk-users] Polycom Park by EFK

On Fri, Dec 3, 2010 at 8:02 PM, Andrew Joakimsen joakim...@gmail.com wrote: Has anyone gotten one-touch call parking to work on Polycom phones via the Enhanced Feature Keys feature working? I've looked at various examples, they appear correct, but the phones (501, 3.1.x firmware) show the Park

[asterisk-users] 1.8 Console Welcome Message

With previous Asterisk versions when running asterisk -r a welcome message is displayed with the version. I just upgraded to 1.8 and noticed it is not appearing. All I get is Verbosity is at least 3 and the console prompt. I looked at main/asterisk.c and still see the welcome message code. Any

Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces

The Loop Back Plug on the link you provided is correct. You take a few inches of CAT5 and remove the outer jacket. Loop the wires into the RJ-45 connector like the diagram shows and then crimp. Ryan On Tue, Oct 5, 2010 at 3:02 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friend

[asterisk-users] Asterisk 1.6.2.12 Download

Anybody else notice that the 1.6.2.12 download has a version and changelog for 1.6.2.12-rc1? http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.12.tar.gz Ryan -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] 'dirty' upgrade of 1.4

When you run make, it compiles the binaries in the src directory. Once it is done compiling stop asterisk. Running make install will copy the compiled binaries into their respective folders on your system. Then just start asterisk. If you need to revert, stop asterisk, run make install in the old

[asterisk-users] Exchange UM Play on Phone

I haven't been successful in getting this to work. The issue looks to be that Asterisk is wanting peer authentication for the invite request as it sends back 401 Unauthorized. I am using FreePBX 2.7 and have tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are type=peer transport=tcp

Re: [asterisk-users] Exchange UM Play on Phone

On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote: I haven't been successful in getting this to work. The issue looks to be that Asterisk is wanting peer authentication for the invite request as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have tested both

Re: [asterisk-users] Integration with Toshiba Strata DK424

On Sat, Jul 24, 2010 at 12:44 PM, Joel Maslak jmas...@antelope.net wrote: I'm posting here in case anyone else runs into this and needs some help. I'll probably update the voip-info Wiki pages on Toshiba integration in a bit.  Asterisk 1.6 makes things a bit easier than what is on that page.

Re: [asterisk-users] Exchange UM Play on Phone

On Sat, Jul 24, 2010 at 12:30 PM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote: I haven't been successful in getting this to work. The issue looks to be that Asterisk is wanting peer authentication for the invite request

Re: [asterisk-users] Exchange UM Play on Phone

On Sat, Jul 24, 2010 at 9:25 PM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jul 24, 2010 at 12:30 PM, Ryan Wagoner rswago...@gmail.com wrote: On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner rswago...@gmail.com wrote: I haven't been successful in getting this to work. The issue looks

Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?

On Fri, Jul 9, 2010 at 4:28 AM, Gilles codecompl...@free.fr wrote: On Mon, 05 Jul 2010 12:45:34 +0200, Gilles codecompl...@free.fr wrote: Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP...

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Tue, Jul 6, 2010 at 10:19 AM, unsero...@aol.com wrote: The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both compile but need to be tested to verify that they work. I have the 1.6.2.9 version in production and plan to put the 1.6.1.20 version in sometime this weekend. In

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 3:26 PM, unsero...@aol.com wrote: -Original Message- From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, Jul 1, 2010 6:19 pm Subject: Re: [asterisk-users] Update

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle supp...@drdos.info wrote: CunningPike wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com  wrote: We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great. There is a much newer patch for 1.4 that can

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info wrote: Ryan Wagoner wrote: together one for 1.4 that compiles. I'll post both to the list hopefully later today. Thank you! Doug -- The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both compile but need

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com wrote: Sounds great. Could you please give me a hint how to install the patch? Sorry for my stupid question but I'm a newbie to Asterisk. Thanks. -Original Message- From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote: Thanks a lot. Applying the patch gives me a Hunk #5 failed at 9881 -Original Message- From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

Re: [asterisk-users] Update the LCD with the callee's name after dialing

On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote: On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote: Thank you Andrew, I will check it out.  We are currently running 1.4. -Matt On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com

Re: [asterisk-users] Codec negotiation

On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! Because the codec is already chosen before the call is made, and you told it that g722 is permitted. There are all sorts of discussions in play about codec negotiation, but at this point

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

On Wed, Jun 23, 2010 at 12:57 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote: completely as well. Below I've posted a patch that responds with a 200 OK to these keep-alive requests, and I believe also solves the temporary loss of

[asterisk-users] Codec negotiation

I have Polycom phones that support the g722 codec. Adding allow=g722 to the [general] section of sip.conf works great and I can make calls between the phones using g722. However Asterisk is negotiating g722 for calls going out my voip provider and transcoding these to ulaw. In sip.conf for the

Re: [asterisk-users] ISP down internal phones become unavailable

On Tue, Jun 22, 2010 at 9:33 AM, Mr Shunz mrsh...@gmail.com wrote: Hi all, I have a PRI, and when the Internet connection goes out so do my phones.  I suspect it is some type of DNS issue.  I do have a SIP trunk, and it appears that if I lose DNS to the SIP trunk, the entire PBX is offline.  

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

On Tue, Jun 22, 2010 at 8:30 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote: On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote: James

[asterisk-users] ISP down internal phones become unavailable

I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan

Re: [asterisk-users] Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)

On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote: James Lamanna schrieb: It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding

Re: [asterisk-users] How to stop intruder from registering sip?

On Sun, Jun 13, 2010 at 4:06 PM, sean darcy seandar...@gmail.com wrote: On 06/13/2010 01:59 PM, Dave Platt wrote: If you leave your asterisk box open to the world with passwords like you deserve to be hacked.. Well, without making a moral judgment, I will agree that you are *going* to

Re: [asterisk-users] Caller ID questions

On Sat, May 22, 2010 at 11:28 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 22 May 2010, GlenM wrote: Hello Folks; I have a dilemma: I have a client with Asterisk 1.4x and he needs to have a record of all incoming calls - caller ID and date/time is sufficient. Since I am

Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?

On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met

[asterisk-users] T.38 Fax With Flowroute SIP Provider

Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in sip.conf. When I receive a fax it tries to negotiate T.38 and Flowroute sends back a Bad Request

Re: [asterisk-users] T.38 Fax With Flowroute SIP Provider

:Body 11 [ 23]: a=T38FaxUdpEC:t38UDPFEC On Thu, May 6, 2010 at 6:54 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully

Re: [asterisk-users] T.38 Fax With Flowroute SIP Provider

On Thu, May 6, 2010 at 7:11 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/06/2010 05:46 PM, Ryan Wagoner wrote: Does anybody have T.38 faxing working with Flowroute? I am running Asterisk 1.6.2.7-rc3

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