[asterisk-users] choppy calls version Asterisk 13.17 on CentOS 7

Hello, A few months ago we upgraded our server from Asterisk 1.8.22.0 on CentOS 5.9 to Asterisk 13.13.1 on CentOS 7. We are still using SIP not PJSIP. Since the upgrade our remote users' conversions are choppy. Here is what my sip.conf looks like for the users with most problems:

[asterisk-users] upgrading asterisk 13.13.1 to latest version best practices

Hello, Best practices examples to upgrade Asterisk 13.13.1 to latest version? Any suggestions? Thanks, Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

X,2,Dial(SIP/voip1/13781${EXTEN:1},80) exten => _7XXX,n,Congestion() exten => _7XXX,n,Hangup() how would I change it? I have look in cdr.conf and logger.conf Thanks, From: Motty Cruz [mailto:motty.c...@gmail.com] Sent: Monday, April 03, 2017 3:52 PM To: 'Aster

[asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID

Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number

[asterisk-users] how to hangup this channel "Message/ast_msg_queu

omega*CLI> core show channels Channel Location State Application(Data) Message/ast_msg_queu 4002@sipphones:2 Up VoiceMail(4002@default,u) "Message/ast_msg_queu" it's been up for the last day, how to hangup this channel? --

Re: [asterisk-users] CDR reporting solution

her MySQL or cdrlite and a quickie sql query? I could > have added postgres, but I'm a DB bigot. That would work too. > > > On 03/22/2017 01:46 PM, Motty Cruz wrote: > >> >> Hello, I am looking for CDR reporting solution? Any suggestions? I am >> using Asterisk 13.

[asterisk-users] CDR reporting solution

Hello, I am looking for CDR reporting solution? Any suggestions? I am using Asterisk 13.13.1 I would like a report on number of calls per extension. Thanks, Motty -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] fail2ban Asterisk 13.13.1

Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"' failed for ‘offending-IP:53911' -

[asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions?

[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)

Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error: -- SIP/voipeer-084b redirecting info has changed,

Re: [asterisk-users] Asterisk 13.13.1

ytle Sent: Monday, January 30, 2017 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 >>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1.

Re: [asterisk-users] Asterisk 13.13.1

source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, c

[asterisk-users] Asterisk 13.13.1

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone

Re: [asterisk-users] T1 -Asterisk server - Analog lines

Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 -Asterisk server - Analog lines On 12/06/2016 06:45 PM, Motty Cruz wrote: Any suggestions on how to convert digital signal to analog? I do this with an ADIT 600 channel bank that I got off of ebaY many years ago. A quick

Re: [asterisk-users] how to add area code to outgoing number in Asterisk 13.13

Thank you Carlos, you’re right I am using PJSIP. Should I not use it? Thanks, Motty From: Carlos Chavez [mailto:cur...@telecomab.mx] Sent: Saturday, December 31, 2016 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Motty Cruz Subject: Re: [asterisk-users] how

[asterisk-users] how to add area code to outgoing number in Asterisk 13.13

Hi, my SIP provider requires 10 digits for all outgoing calls; Users dial 7 digits for outgoing. Here is how I added the area code to all outgoing calls in Asterisk 1.8 Extensions.conf ; Adding Area code and striping 7 for local numbers exten => _7XXX,n,Set(CALLERID(all)="My ID" )

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

Yves, I have a SoundStation IP 6000: My sip.conf [1006] type=friend context=sip-phone call-limit=1 trustrpid=no callerid="Conference Room3" <1006> disallow=all allow=ulaw allow=alaw username=1006 secret=secret1 dtmfmode=rfc2833 host=dynamic mailbox=1000 nat=yes canreinvite=no

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

can you provide the configuration on sip.conf file? Do you have the following settings under the account number or ext number? host=dynamic nat=yes for instance my configuration sip.conf file is as follow: [1005] type=friend context=sip-phone call-limit=1 trustrpid=no

[asterisk-users] T1 -Asterisk server - Analog lines

Hello All, The problem I'm facing is the following: two machines that require analog signal. However, connection to the world is setup as follow: T1 connection to Asterisk server Any suggestions on how to convert digital signal to analog? Type of card on my Asterisk server is wct4xxp

Re: [asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers

er. Thanks, Mottyh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, November 06, 2016 12:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 to Asterisk 1

[asterisk-users] Asterisk 1.8 to Asterisk 13.11 appending area code to local numbers

Hello, I would like to add area code to local numbers, it worked like a charm on Asterisk 1.8 but does not work on Asterisk 13.11. Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to local numbers exten => _9XXX,n,Set(CALLERID(all)="$CallerID" <3818008000>) exten =>

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Thank you for your help! Centos 7 firewall was enable. systemctl stop firewalld issue fixed. Thanks, On Thu, Oct 13, 2016 at 3:54 PM, Victor Villarreal wrote: > Ok. > > Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of > the Polycom hardphone.

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

to find some clue ir send me back some trace. Cheers. El oct. 13, 2016 1:45 PM, "Motty Cruz" <motty.c...@gmail.com> escribió: Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend

[asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas?

[asterisk-users] Asterisk Secure SIP session TLS port 5061

I finally secure SIP session between Asterisk server and a remote client. My questions is the following; do I need to open port 5061 UDP on my firewall or just port 5061 TCP for SIP sessions.? I am not interested in securing RTP only SIP sessions. Thanks for your help! --

Re: [asterisk-users] Asterisk 1.8 secure SIP session only

1 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 secure SIP session only Your CA cert is missing. Add in sip.conf: tlscafile=/etc/asterisk/keys/ca.crt You don't need: tlscapath=/etc/asterisk/keys On 4 May 2016 at 19:43, Motty

[asterisk-users] Asterisk 1.8 secure SIP session only

Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I keep getter an error, == Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection: FILE * open

[asterisk-users] dahdi auto-call multiple destinations

Am trying to get a script to call multiple destinations, for instance I have the following file: Channel: dahdi/g1/6078880 CallerID: "Room Tempeture" <800579> MaxRetries: 2 RetryTime: 60 WaitTime: 20 This works great for me, however I am trying to add a secondary number, is that possible to

[asterisk-users] Asterisk 1.8.22.0 built - encrypt authentication

Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Asterisk 1.8.22.0 built - encrypt authentication

, Oli-net wrote: Hello, Try MD5SUM like that in a terminal echo -n myverysecretpassword | md5sum It will return you something like that 22ea6cf875d66b15d275684427275dfdf witch is your password in an MD5 format. Hope this help Oliver Le 29/07/2015 16:30, Motty Cruz a écrit : Hello, I would

Re: [asterisk-users] Sending E-Mail from voicemail with AND without attachment

Hello, this worked for me: 500 = mysecret,Motty mailbox,mo...@domain.com,,tz=pacific,attach=yes|delete=1 Thanks, Motty On 07/10/2015 07:45 AM, Luca Bertoncello wrote: Hi again, I'm trying to send two E-Mails when a message comes in the voicemail, the first WITH the attachment, and the

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

would like to add background music if authentication failed, then after 6 minutes hangup any ideas, suggestions? On 07/07/2015 09:09 AM, Motty Cruz wrote: Hello, I used this guide, it worked for me: http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html Thanks, On 07/06/2015 04

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

+Application_Authenticate You can either give it a single PIN to use for all calls, Authenticate using a value in the Asterisk Database, Or use a plain text file for the PIN's On Mon, Jul 6, 2015 at 2:43 PM, Motty Cruz motty.c...@gmail.com mailto:motty.c...@gmail.com wrote: Hello All, I will like

[asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

Hello All, I will like to configure Asterisk to use PIN Code for all outgoing international calls. Also, any suggestions as to when should I prompt users for code prior to dialing the number or after dialing the number? can someone provide with a example on how to accomplish this goal? I

[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user

Hello, I would like to setup a mechanism to trigger an alarm if user is deal too many numbers within a very short period of time. Safeguard against users hacked accounts. can someone help? Thanks, -- _ -- Bandwidth and

Re: [asterisk-users] adding area code

this code worked for me, here is what I did and worked for me: exten = 1381+NXX,1,Set(CALLERID(number)=3817383444) exten = 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) Thanks for you help! On 04/27/2015 02:56 PM, Matt Riddell wrote: On 27Apr, 2015, at 16:39, Motty Cruz motty.c

Re: [asterisk-users] adding area code

: On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz motty.c...@gmail.com wrote: here is what I did and worked for me: exten = 1381+NXX,1,Set(CALLERID(number)=3817383444) exten = 1+NXXNXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have

Re: [asterisk-users] adding area code

Thanks for your reply, [globals] AREACODE=381 [outbound] exten = _NXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: On 27 April 2015 21:32:42 BST, Motty Cruz motty.c...@gmail.com wrote: Hello, I

Re: [asterisk-users] adding area code

and the last 7 digits of your dialed phone number exten = _9XXX,n,Dial(SIP/${dialnumber},35) Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: Motty Cruz motty.c...@gmail.com *Sent*: Monday, April 27

[asterisk-users] adding area code

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty --

Re: [asterisk-users] adding area code

forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. Thanks, On 04/27/2015 02:38 PM, Motty Cruz wrote: here is what I have: exten = _9XXX,1,Set(l_HomeAreaCode=381) exten = _9XXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten = _9XXX,n,Dial(SIP/SIP-Provider

[asterisk-users] how to strip +1 out of incoming number

Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is there a way to strip +1 out of caller ID? -- Thanks for your support, Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

Thank you AJ, I will certainly not copy and past; I want to believe I understand the risk. I needed some kind of direction, thank you for your support. -Motty On Fri, Sep 19, 2014 at 2:51 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 18 Sep 2014, motty cruz wrote: Hello, I

[asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty --

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

Thank you Julian, would it be possible to block calls to international calls except certain countries? I just want to make sure that if attackers try to place calls outside the states they not succeed. Thanks, Motty On Thu, Sep 18, 2014 at 12:55 PM, Julian Beach jb_s...@trink.co.uk wrote:

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

...@lists.digium.com] *On Behalf Of *motty cruz *Sent:* Thursday, September 18, 2014 4:52 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism Thank you Julian, would

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

-Basics-SECT-3.6 is not helpful to you. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *motty cruz *Sent:* Thursday, September 18, 2014 5:27 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk

[asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty --

Re: [asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

Thanks Eric, for point to polycom instructions I will give it a try. Kevin, I am sure called is not originating from our system, Thanks for your support. On Tue, Sep 16, 2014 at 9:08 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: Hello, a user outside the office regularly gets a

Re: [asterisk-users] chan_sip.c:23647 handle_request_invite: Failed to authenticate device

Hello Deepak, 601sip:601@111.118.185.107;tag=2f498fbd is the 111.118.185.107 your server IP? it could be a client trying to authenticate as Rusty suggest or an attacker attempting to gain access, if you want to find out what IP address that request is coming from do the following command. make

[asterisk-users] Asterisk failed to authenticate device - attack attempt.

Hi all, I continue to see the following msg on my Asterisk log: [Sep 8 15:34:37] NOTICE[7375]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 9009sip:9...@196.107.xx.xx;tag=8dd48dd2 IP: 196.107.xx.xx is my asterisk server IP address. I don't know what it means and how to

[asterisk-users] Asterisk secure fine tune - stop attack

Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Thanks in

Re: [asterisk-users] Asterisk secure fine tune - stop attack

...@ovm-group.com wrote: Am 04.09.2014 16:44, schrieb motty cruz: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422

Re: [asterisk-users] Asterisk secure fine tune - stop attack

, Sep 4, 2014 at 8:19 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 04 Sep 2014, motty cruz wrote: Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs

Re: [asterisk-users] Asterisk secure fine tune - stop attack

Thank you all for your support, your suggestions are welcome. Thanks, On Thu, Sep 4, 2014 at 9:26 AM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 4/9/14 4:58 pm, Eric Wieling wrote: If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses

[asterisk-users] share mailbox Asterisk 1.8.22

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten = 102,hint,SIP/${EXTEN} exten = 102,1,Dial(SIP/101SIP/102,20,t)

Re: [asterisk-users] share mailbox Asterisk 1.8.22

checks the voicemail, do they hear the correct voicemails? Ours clears just fine in this situation. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 04:37:26 PM: From: motty cruz motty.c...@gmail.com To: Asterisk

Re: [asterisk-users] share mailbox Asterisk 1.8.22

Thanks Kevin, can you provide me with example of your code? if you don't mind? Thanks, On Tue, Jun 24, 2014 at 3:46 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: asterisk-users-boun...@lists.digium.com wrote on 06/24/2014 05:36:16 PM: From: motty cruz motty.c...@gmail.com

Re: [asterisk-users] Asterisk 1.8.22

:) -- *From:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com *Sent:* Monday, May 12, 2014 5:43 PM *To:* Asterisk Users List *Subject:* [asterisk-users] Asterisk 1.8.22 Hello, recently I have seen spike in attacks on my

[asterisk-users] Asterisk 1.8.22

Hello, recently I have seen spike in attacks on my asterisk server, this is what I get on the LCD of my phone: 201@76.220.5.205 or calls from 1000 sip1000@76.2230.5.205, have any idea on how to stop this calls? Thanks, -- _ --

[asterisk-users] how to hangup Local/100 channel

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i@autoatten:2] Goto(Local/100@sipphones-01b2;2, s,2) in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on

Re: [asterisk-users] how to hangup Local/100 channel

: On Mon, 5 May 2014, motty cruz wrote: one of the extensions fall into a loop, I don't know how to hangup that channel -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200@sipphones-01b2;2 any ideas? If you're asking how to prevent

Re: [asterisk-users] Asterisk 1.6

. Since all out clients are under our control we use openvpn a lot and yealink and other phones have it built in so they can connect directly once initially setup Cheers Duncan On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote: that sounds feasible, Thanks Michelle, On Fri

[asterisk-users] Asterisk 1.6

Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673 handle_request_register: Registration from '4941 sip:4941@public_ip' failed for '194.100.46.132 194.100.46.132:56714' - Wrong password [Apr 4 06:56:00] NOTICE[21745]: chan_sip.c:25673

Re: [asterisk-users] Asterisk 1.6

setup. How many sip phones do you have outside your network? If few and in well-known IPs, consider limiting access to only those (and the sip provider you are using). On 04/04/2014 09:00 AM, motty cruz wrote: Hello All, my asterisk server is constantly under attack [Apr 4 06:56:00

Re: [asterisk-users] Asterisk 1.6

absolutely right A J, thanks for the heads up. I do not intent to implement that solution in production server, I hope to learn it first, build a test server and monitor for a few days or weeks. Thanks again, On Fri, Apr 4, 2014 at 7:38 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On

Re: [asterisk-users] Asterisk 1.6

Hello Ishfaq, outside users usually travel around the country and connect from different network, so it won't be possible to lock it down to specific IP. Thanks for your support. On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 4 April 2014 15:22, motty cruz

Re: [asterisk-users] Asterisk 1.6

:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of motty cruz motty.c...@gmail.com *Sent:* Friday, April 4, 2014 11:15 AM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Asterisk 1.6 Hello Ishfaq, outside users usually travel around

Re: [asterisk-users] callerid overwrite

look like the issue continues, I am unable to overwrite callerid from sip.conf in extensions.conf, In sip.conf under [general] trustrpid = no should i change it to yes? Thanks On Tue, Jan 28, 2014 at 1:06 PM, motty cruz motty.c...@gmail.com wrote: Thank you for your reply, I updated

[asterisk-users] callerid overwrite

Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be mycompanyinc but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid=iuser 101 disallow=all allow=ulaw

Re: [asterisk-users] callerid overwrite

Thank you for your reply, I updated extensions.conf file to reflect your suggestion, I will monitor Asterisk for any more issues, Thanks, On Tue, Jan 28, 2014 at 11:23 AM, Andres and...@telesip.net wrote: On 1/28/14, 1:55 PM, motty cruz wrote: Hi all, I'm having issues with overwrite

[asterisk-users] Asterisk 1.8.22.0 Polycom ip soundpoint sp450

Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty --

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

number is dark same as the background so that means is not subscribing to the Asterisk server. Thank you very much. On Thu, Jan 2, 2014 at 8:19 AM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: asterisk-users-boun...@lists.digium.com wrote on 01/02/2014 10:03:19 AM: From: motty cruz

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

Thank you all, After setting the phone to factory defaults, entered configuration parameters, phone is working again. I really don't know why all sudden stop working. at least know i have a working phone I will go thoroughly through the logs, I hope to find the answer, if I do I will post it

[asterisk-users] Asterisk 1.8.22

Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks, --

Re: [asterisk-users] Asterisk 1.8.22

. Mitul On Friday, November 8, 2013, motty cruz wrote: Hello, I have a fully functional Asterisk Server, I want to configure this server to be able to process call from Skype, can someone point me to a howto? or if there are suggestions on best way to approach this problem. Thanks

[asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104@sipphones:1]

Re: [asterisk-users] asterisk 1.8.22 - : app_meetme.c:1248 build_conf: Unable to open DAHDI pseudo devic

Thanks Johan, I did noticed /etc/dahdi so you're right it was installed on one point, I re-install dahdi and problem went away. Thank you very much! On Thu, Jun 6, 2013 at 1:45 PM, Johan Wilfer li...@jttech.se wrote: 2013-06-06 22:21, motty cruz skrev: Hello All, I upgraded Asterisk

[asterisk-users] monitoring Asterisk 1.8

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- Bandwidth and

Re: [asterisk-users] monitoring Asterisk 1.8

in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm

Re: [asterisk-users] monitoring Asterisk 1.8

/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios

Re: [asterisk-users] asterisk 1.8.10.1 meetme

Rose jr...@digium.com wrote: motty cruz wrote: Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference

[asterisk-users] asterisk 1.8.10.1 meetme

Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen

Re: [asterisk-users] Polycom phones and ring no answer/302 Moved Temporarily

I have Polycom IP550. The Forward No Answer is working fine when enabled. I was looking at the sip.cfg but don't know exactly what to look for, can you give me a hint to where would i find that option? Thanks, On Wed, Dec 12, 2012 at 1:48 PM, Justin Sherrill justin.sherr...@americanrocksalt.com