[asterisk-users] Polycom getting DCHP address from wrong VLAN

Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config th

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 7

Hi, I cannot get asterisk to start again after the g729 install failed. kindly advise what's the problem. Thank's -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a liv

[asterisk-users] Radius client support

Hi Will radius client in asterisk can use with third party radius servers instead of freeradius ?,if supports how do I configure asterisk to make it work. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http:

Re: [asterisk-users] Dahdi error

On 10/7/10 2:07 PM, Flavio Miranda wrote: > asterisk:/etc/asterisk# /etc/init.d/dahdi start > Loading DAHDI hardware modules: > FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): > Device or resource busy > wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp:

[asterisk-users] RES: Alert-Info advice

Hi, I use it on Linksys PAP2, I think you need to put brackets after Alert-info on your code. Check mine: exten => _1XXX,1,SIPAddHeader(Alert-Info: ) Atenciosamente, Rafael Prado PRACTIS - Comunicação & Tecnologia Av Aquidaban, 766 - Conj 51 CEP 13026-510, Campinas/SP - Brasil

[asterisk-users] REINVITE with Auth Credentials has different SDP Codec

Hi I have a call from Service Provider (SP) to Asterisk to User User sends a T38 REINVITE Asterisk passes that to SP SP challenges the INVITE Asterisk sends INVITE with credentials but sends G711ulaw in the SDP instead of T38 udptl... Obviously Fax fails.. Any ideas on how I can maintain th

[asterisk-users] asterisk router

Looking for a router to connect to a 5/50 cable modem that works with Sip. A Crisco RVS4000 installed now has real problems with Sip, one-way audio and throughput not up to the WAN speed. No VPN needed, something affordable, $200-$350 US range. Every thing I looked at in that range had some r

[asterisk-users] Asterisk 1.8.0 Release Candidate 3 Now Available

The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process.

Re: [asterisk-users] integrate Intertel Axxess with Asterisk

On Wed, Oct 6, 2010 at 5:00 PM, marvin horst wrote: > Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone > system via a SIP trunk using the IPRC card? I have, believe it or not, integrated Asterisk with Inter-Tel. However, not via SIP. Run the costs. When I did, it was w

Re: [asterisk-users] ADA: DOA?

On 8/10/10 6:02 AM, Danny Nicholas wrote: > FWIW, "open source" is only "truly dead" when you can't find anywhere to > download the source. It wasn't ever Open Source, and source was never provided. I checked a while ago and that's never likely to happen, so yep (at least as of last time) it is

[asterisk-users] Dahdi error

Hi all, What hell hapen here? asterisk:/etc/asterisk# /etc/init.d/dahdi startLoading DAHDI hardware modules:FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done w

Re: [asterisk-users] Checking SIP Headers existence and content

Favorites? voip-info.org should be your homepage. On Thu, Oct 7, 2010 at 9:26 AM, Administrator TOOTAI wrote: > Le 05/10/2010 05:13, VoIP Question a écrit : > > Hello, > > Hi > > > > > I would like to verify if a specific SIP header exists, and if yes, > > extract the partial content from anothe

Re: [asterisk-users] ADA: DOA?

On Thu, October 7, 2010 1:02 pm, Danny Nicholas wrote: > FWIW, "open source" is only "truly dead" when you can't find anywhere to > download the source. I *totally* agree... if you can find me the source. I have, at this moment, at least, no reason to believe ADA is OSS -- indeed, looking at it,

Re: [asterisk-users] ADA: DOA?

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, October 07, 2010 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ADA: DOA? 2010/10/7 Paul Hayes

[asterisk-users] Voice drop out

Hi, I am facing some voice drop in inbound, outbound, and IVR. But while checking the process of the CPU and memory utilization is very less. Mem: 21304K used, 36500K free, 0K shrd, 1896K buff, 13228K cached The voice drop is in systemic. I am not too sure what to check... all the configuration

Re: [asterisk-users] ADA: DOA?

2010/10/7 Paul Hayes > On 06/10/10 20:25, Ken D'Ambrosio wrote: > > Hey, all. While ADA can still be downloaded, that's about all that I > see. > > No development, no recent mention, and -- perhaps worst of all -- it > > appears not to work properly under 64-bit systems. So, assuming Digium's

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On Thu, Oct 07, 2010 at 02:57:27PM +0200, Jonas Kellens wrote: > nat=yes is set as a global parameter and also in the realtime MySQL > sip_buddies database I have for every peer nat=yes. > > I then find it very strange that when placing these Snom phones in my > environment (for configuration) w

Re: [asterisk-users] AMI getting related channels in Ringing state

On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote: > Issuing the AMI Status command results in a list of active channels. But > how to figure out which channels are related before the call is > answered? Anybody? My workaround for this problem is setting a persistent variable in the

Re: [asterisk-users] RTP Read too short

On 10/07/2010 10:36 AM, Bryant Zimmerman wrote: > Hi All > > In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too > short > > I get these all of the time things seem to be working fine but I am > trying to figure out if there is a way to resolve these Warnings. > I am running

Re: [asterisk-users] SIP authentication - Thoughts please

On 7 October 2010 10:10, Stefan Schmidt wrote: > Am 07.10.10 10:52, schrieb Steve Davies: >> Hi, >> > > > Hello, > > i just want to say something about point 4 which comes to my mind about > security. > >> >> 4) I am not sure whether it is worth dropping through and testing auth >> against other

Re: [asterisk-users] ADA: DOA?

On 06/10/10 20:25, Ken D'Ambrosio wrote: > Hey, all. While ADA can still be downloaded, that's about all that I see. > No development, no recent mention, and -- perhaps worst of all -- it > appears not to work properly under 64-bit systems. So, assuming Digium's > abandoned it, are there any su

[asterisk-users] RTP Read too short

Hi All In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too short I get these all of the time things seem to be working fine but I am trying to figure out if there is a way to resolve these Warnings. I am running asterisk 1.6.2.13 Any direction is appreciated. Thanks Brya

[asterisk-users] How to change features.conf's atxfer dialing tone ?

Hi, I'm facing the following request : "When someone is starting an assisted transfer using Asterisk's features codes, he will ear a prompt saying "Transfer" and then a dialing inviting him to dial the number he tries to reach. This tone volume is qualified as a bit too load." Is it possible to c

Re: [asterisk-users] Checking SIP Headers existence and content

Le 05/10/2010 05:13, VoIP Question a écrit : > Hello, Hi > > I would like to verify if a specific SIP header exists, and if yes, > extract the partial content from another header. > > 1. Is there a way to verify if a specific header exists? > 2. How do I extract data that is between the first :

[asterisk-users] convert g729A-g729B and vice-versa

Hi all. Is there a free, or at least non-expensive, solution that can convert g729A <-->g729B (with VAD)? The no-support for g729B on Asterisk gives me a BIG headache… Thanks, Harel This electronic message and any files transmitted with it are confidential and i

Re: [asterisk-users] Alert-Info advice

I use the following syntax for sipura i think, and it works fine for me. exten=> s,1010,SipAddHeader(Alert-Info: ) exten=> s,1020,SipAddHeader(Alert-Info: ) exten=> s,1030,SipAddHeader(Alert-Info: ) exten=> s

Re: [asterisk-users] Polycom: full caller ID?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio Sent: Thursday, October 07, 2010 9:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom: full caller ID? Hi, all. When I get

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On 10/07/2010 04:18 PM, Philipp von Klitzing wrote: > Hi! > > >> I'm having difficulty with registering a SIP account in a Snom 320 IP- >> phone. >> > Do a SIP trace on your SNOM phone, and you will most probably see that > the 401 reply of Asterisk does not arrive on the phone. Then chec

[asterisk-users] Polycom: full caller ID?

Hi, all. When I get calls on my SoundPoints, I only see the number -- is there a way to get the alpha portion of the CID, as well? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- ___

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

Hi! > I'm having difficulty with registering a SIP account in a Snom 320 IP- > phone. Do a SIP trace on your SNOM phone, and you will most probably see that the 401 reply of Asterisk does not arrive on the phone. Then check your STUN/ICE settings on the phone in combination with the nat= settin

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On 10/07/2010 02:36 PM, Daniel Tryba wrote: > On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote: > >>> It's the same account, the same password, but other agent. >>> >>> Can anyone help me with this please ?! I see no difference but there >>> must be !! >>> >> The difference

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On 10/07/2010 02:36 PM, Daniel Tryba wrote: > On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote: > >>> It's the same account, the same password, but other agent. >>> >>> Can anyone help me with this please ?! I see no difference but there >>> must be !! >>> >> The difference

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On 10/07/2010 02:36 PM, Daniel Tryba wrote: > On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote: > >>> It's the same account, the same password, but other agent. >>> >>> Can anyone help me with this please ?! I see no difference but there >>> must be !! >>> >> The difference

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

Hello ! Thank you very much for your quick answer !! nat=yes is set as a global parameter and also in the realtime MySQL sip_buddies database I have for every peer nat=yes. I then find it very strange that when placing these Snom phones in my environment (for configuration) work very well, a

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On Thu, Oct 07, 2010 at 02:24:59PM +0200, Daniel Tryba wrote: > > It's the same account, the same password, but other agent. > > > > Can anyone help me with this please ?! I see no difference but there > > must be !! > > The difference is the SNOM is using rport and Zoiper isn't. Is nat for > th

Re: [asterisk-users] Difference

Thanks for sharing all of your thoughts and information. If anyone knows a good article about asterisk 1.8 then please let me know about it. I have read the presentation by Kevin Fleming but more information is always good. Cheers On Wed, Oct 6, 2010 at 10:28 AM, Miguel Molina wrote: > I find 1

Re: [asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

On Thu, Oct 07, 2010 at 01:54:58PM +0200, Jonas Kellens wrote: > It's the same account, the same password, but other agent. > > Can anyone help me with this please ?! I see no difference but there > must be !! The difference is the SNOM is using rport and Zoiper isn't. Is nat for this client set

[asterisk-users] 401 Unauthorized with Snom but not with Zoiper softphone

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200

Re: [asterisk-users] How to test BRI lines energy saving mode ?

Well, to go slightly O/T: If you read the issue tracker for 17270 - it appears to be a LibPri 'fault'. So I would say that the main work would need to be in LibPri . Maybe someone who knows LibPri and DAHDI better can explain how the two combine... Cheers Andy -Original Message- From:

Re: [asterisk-users] Problems compiling Asterisk on Debian

On Mon, Sep 27, 2010 at 06:21:16PM +0200, Danny Dias wrote: > Thanks Dean, > > I've done it before, that's why i'm here asking :( take a look: > > r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# apt-cache search > linux-headers-$(uname -r) > linux-headers-2.6.26-2-amd64 - Header files for Linu

Re: [asterisk-users] How to test BRI lines energy saving mode ?

2010/10/7 Andrew Thomas > The D-channel isn't actually 'dropped' - it is put in to a 'power-save' > state. > > See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to > 'Activation / Deactivation' for more information. > > Anyway - this is a known 'problem' - > https://issues.asterisk

Re: [asterisk-users] SIP authentication - Thoughts please

Am 07.10.10 10:52, schrieb Steve Davies: > Hi, > Hello, i just want to say something about point 4 which comes to my mind about security. > > 4) I am not sure whether it is worth dropping through and testing auth > against other peers if there is no username match. Can auth ever > succeed und

[asterisk-users] SIP authentication - Thoughts please

Hi, We have a scenario where we need multiple discrete SIP trunks (peers) from/to a single endpoint. Because the authentication system starts by matching IP address, it only ever matches on one of the SIP peer entries, and ignores the others. This is documented behaviour and as such is "correct".

[asterisk-users] Fw: asterisk > cisco gateway > westell > isdx

Anyone? - Forwarded Message Hi, I am hoping someone can help me with a problem I am having. I am trying to setup a connection from an Elastix 2 server to a Siemens isdx PBX. The setup is as follows Elastix 2 *sip trunk* Cisco 2621XM router with 2 E1 voice interfaces *QSIG* W

Re: [asterisk-users] How to test BRI lines energy saving mode ?

The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information. Anyway - this is a known 'problem' - https://issues.asterisk.org/view.php?id=17270 As there is no f

Re: [asterisk-users] How to test BRI lines energy saving mode ?

2010/10/7 Lyle Giese > Olivier wrote: > > Hello, > > > > If my understanding is correct, these days it seems that many ISDN BRI > > lines are configured in energy saving mode in which signalling > > D-channel is "dropped" until a new call comes in. > > > > Is it possible to replicate this behavio