Re: [asterisk-users] help with crash

Hello Federico, Can you please review the Bug Report requirements, and submit a new bug report for this issue? https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/ Also Note: Before filing a bug report... Your issue may not be a bug or could have been fixed already. Run

[asterisk-users] help with crash

2023-11-08 18:14:13] ERROR[571246][C-17e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

On Wed, 26 May 2021, Jonathan H wrote: AGI Rx << SET AUTOHANGUP 5 AGI Tx >> 200 result=0 AGI Tx >> HANGUP << This does raise a question in my mind... The AGI protocol is: your AGI sends a request (the Rx line) and receives a response (the Tx line). 1 line out, 1 line in. If the 'HANGUP'

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

On Wed, 26 May 2021, Jonathan H wrote: It just causes AGI to send "HANGUP" and any audio to stop playing. It does NOT hangup the channel, or even send any SIP event. The line just goes silent. I wouldn't expect the AGI() application to send a SIP event. The AGI() application does not care

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

I think I can confidently say, after most of a day and reading the following https://stackoverflow.com/questions/66768885/why-doesnt-asterisk-17-catch-hangup-request-from-pjsip-client-solved https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

On Wed, May 26, 2021 at 1:58 PM Jonathan H wrote: > I have also tried configuring pjsip wizard like this. > > endpoint/rtp_timeout=5 > > And I see this shortly after the "hangup" command has been sent, so > that part is working: > > [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150 >

Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

I have also tried configuring pjsip wizard like this. endpoint/rtp_timeout=5 And I see this shortly after the "hangup" command has been sent, so that part is working: [May 26 17:36:37] NOTICE[1276]: res_pjsip_sdp_rtp.c:150 rtp_check_timeout: Disconnecting channel

[asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+AGICommand_set+autohangup "Cause the channel to automatically hangup at time seconds in the future" SET AUTOHANGUP TIME Looks great. Except... it doesn't. It just causes AGI to send "HANGUP" and any audio to stop playing. It does NOT hangup

Re: [asterisk-users] Help needed with ARI RTP externalMedia bridging please

You're a genius, sir! I don't know how I missed the part about ports, but anyway... Looks for "channelvars": { "UNICASTRTP_LOCAL_PORT": "*14880*", and then vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout '#transcode{vcodec=none,acodec=*a*

Re: [asterisk-users] Help needed with ARI RTP externalMedia bridging please

On Sun, Jan 3, 2021 at 4:14 PM Jonathan H wrote: > Very simply, I want to pipe some external audio into a channel (bridge) > using the externalMedia channel option. > Running Asterisk 18 on ubuntu, here's what I did to try and test things > out: > > open a console tab > vlc -vvv

[asterisk-users] Help needed with ARI RTP externalMedia bridging please

Very simply, I want to pipe some external audio into a channel (bridge) using the externalMedia channel option. Running Asterisk 18 on ubuntu, here's what I did to try and test things out: open a console tab vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 25/12/2020, at 3:08 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > My final issue has been resolved. Very well done Merry Xmas Cheers Duncan > > Please refer to the following post. > > Post: Addendum to Teo En Ming's Guide to Configuring Asterisk/FreePBX

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Duncan Turnbull, My final issue has been resolved. Please refer to the following post. Post: Addendum to Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones Link: http://lists.digium.com/pipermail/asterisk-users/2020-December/295590.html Thank you very much.

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > It is a newly created PJSIP extension with default settings. I have never > configured Do Not Disturb settings before. > > Could it be something else? > > Do I need to run tcpdump on the

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Duncan Turnbull, It is a newly created PJSIP extension with default settings. I have never configured Do Not Disturb settings before. Could it be something else? Do I need to run tcpdump on the Asterisk PBX server again? On 2020-12-24 18:55, Duncan Turnbull wrote: On 24/12/2020, at 6:39

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > I have finally managed to get my Cisco 7960 IP phone to register on my > Asterisk PBX appliance on Christmas Eve 2020. > > You can read my guide here: > > Guide: Teo En Ming's Guide to

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi! What is sngrep? I have never heard of it before. Merry Christmas 2020! On 2020-12-24 13:06, Steve Edwards wrote: On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote: 3. secret is 8 char only, must be numeric My my SIP.cnf file from 2007 contains: image_version:

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Duncan Turnbull, I have finally managed to get my Cisco 7960 IP phone to register on my Asterisk PBX appliance on Christmas Eve 2020. You can read my guide here: Guide: Teo En Ming's Guide to Configuring Asterisk/FreePBX with Cisco 7960 IP Phones Link:

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote: 3. secret is 8 char only, must be numeric My my SIP.cnf file from 2007 contains: image_version: P0S3-8-12-00 line1_password: 346cc89a2526255839534c22ad7790c and my notes say my 9760

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Turritopsis I think the key point maybe making sure the password doesn’t exceed the capacity of the phone. So an 8 char password is a good idea I would be surprised if pjsip doesn’t work but I haven’t tried it with a Cisco phone Whatever gets you working is what you want Have a wonderful

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Thank you for your replies, Duncan Turnbull. I am going to run tcpdump on my Asterisk PBX server. By the way, I found a Youtube video. Youtube video: Cisco 7942g IP Phone Configuration on FreePBX In-Depth(Without Endpoint Manager) Link: https://www.youtube.com/watch?v=gk6w8O3fZlc=youtu.be

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

 Sent from my iPad > On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > You can watch my Youtube video of my Cisco 7960 IP phone. > > The link is: https://www.youtube.com/watch?v=ip_F08jmmio > > My Youtube video shows the Network Configuration

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi Duncan Turnbull, You can watch my Youtube video of my Cisco 7960 IP phone. The link is: https://www.youtube.com/watch?v=ip_F08jmmio My Youtube video shows the Network Configuration settings, SIP Configuration settings and Status of my Cisco 7960 IP Phone. Did you see anything wrong?

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi there > On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Good morning Duncan Turnbull, > > I have posted my Asterisk PBX server debugging output previously in my > original post. The link is: > >

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Good morning Duncan Turnbull, I have posted my Asterisk PBX server debugging output previously in my original post. The link is: http://lists.digium.com/pipermail/asterisk-users/2020-December/29.html I saw many REGISTER requests. Are these REGISTER requests from my Cisco 7960 IP phone?

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi there That answer includes using tcpdump to check for SIP packets and examine the register packet. At this point you have no SIP packets coming from your phone so you are not upto that stage yet. You need to know why there are no SIP packets coming. My guess is your config files have a typo

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Good day from Singapore, I seem to have found the solution at FreePBX community forums. Please check out the following discussion thread. Discussion Thread: Cisco 7940 registration problem RESOLVED Link: https://community.freepbx.org/t/cisco-7940-registration-problem-resolved/30285 But I

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi there I would normally highlight the part but the email is so long I thought I would just note what I can see It appears the Cisco is downloading files. None of the SIP traces show the IP of the phone of the extension Your proxy is at 192.168.1.9 Your phone is at 192.168.1.130 These are

[asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Subject: HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI Good day from Singapore, My Asterisk version: 16.13.0 My FreePBX version: 15.0.16.81 On 7 December 2020, I was able to get Bria softphone to work with my Asterisk PBX

Re: [asterisk-users] Help missing

What is the application that you are missing? On Sun, May 17, 2020 at 01:32 Saint Michael wrote: > I want to see the help when I type core show application , and it's > not available. This is asterisk 16 from sources. I have libxml2-dev > installed. Ubuntu 19 > What am I missing? > Philip >

[asterisk-users] Help missing

I want to see the help when I type core show application , and it's not available. This is asterisk 16 from sources. I have libxml2-dev installed. Ubuntu 19 What am I missing? Philip -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Help with FUNC_MATH

I found my mistake. I was running execif on the result. I needed to change: ExecIf(${MATH(${HOUR_SELECTED}<11)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)})) TO: ExecIf($["${MATH(${HOUR_SELECTED}<11)}" == "TRUE"]?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)})) On Thu, Feb 13, 2020

Re: [asterisk-users] Help with FUNC_MATH

-Commercial Discussion Subject: Re: [asterisk-users] Help with FUNC_MATH John, That is correct. I am trying to figure out why Asterisk is executing the set part of the execif, if it's coming back as false. On Thu, Feb 13, 2020 at 2:10 PM John Kiniston wrote: My Apologies Dovid, I

Re: [asterisk-users] Help with FUNC_MATH

John, That is correct. I am trying to figure out why Asterisk is executing the set part of the execif, if it's coming back as false. On Thu, Feb 13, 2020 at 2:10 PM John Kiniston wrote: > My Apologies Dovid, I think I misunderstood your request. > > You don't have the time you need to

Re: [asterisk-users] Help with FUNC_MATH

in a comparison. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Thursday, February 13, 2020 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with FUNC_MATH HOUR_SELECTED is going to be 1

Re: [asterisk-users] Help with FUNC_MATH

My Apologies Dovid, I think I misunderstood your request. You don't have the time you need to convert in the format of date string, Instead you have your users entering via DTMF when they want something to happen? On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender wrote: > John, > > From looking at

Re: [asterisk-users] Help with FUNC_MATH

sk-users [mailto:asterisk-users-boun...@lists.digium.com] *On > Behalf Of *Dovid Bender > *Sent:* Thursday, February 13, 2020 4:47 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Help with FUNC_MATH > > > > Hi, > > > > I hav

Re: [asterisk-users] Help with FUNC_MATH

Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help with FUNC_MATH Hi, I have some dialplan code that is trying to convert 12 hour time with AM/PM to 24 hour format. The code has something like this: Exten => 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SE

Re: [asterisk-users] Help with FUNC_MATH

John, >From looking at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John

Re: [asterisk-users] Help with FUNC_MATH

Try using the STRFIME function instead of doing this by hand. https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME *%H* The hour as a decimal number using a 24-hour clock (range 00 to 23). *%I* The hour as a decimal number using a 12-hour clock (range 01 to 12). On Thu, Feb 13, 2020

[asterisk-users] Help with FUNC_MATH

Hi, I have some dialplan code that is trying to convert 12 hour time with AM/PM to 24 hour format. The code has something like this: Exten => 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)})) Earlier on in the dialplan HOUR_SELECTED is set to 12. When

[asterisk-users] Help migrating voicemail to database

I am in the process of configuring my systems to store voicemail in a mysql databse as opposed to on the filesystem, as it is now. My backup server is currently configured for db storage, while my production server is still using the filesystem during testing. When I record a vm message on my

Re: [asterisk-users] Help me please i am facing much trouble

On Sat, 16 Jan 2016, waqas.mehmood90 wrote: How to get user extension number in agi php scrip from which he's calling on ivr i am using cid and able to get his name but not his extension no please help me thanx in advance You can use the 'agi set debug on' CLI command to enable AGI

[asterisk-users] Help me please i am facing much trouble

How to get user extension number in agi php scrip from which he's calling on ivr i am using cid and able to get his name but not his extension no please help me thanx in advance Sent from my Samsung Galaxy smartphone.-- _

Re: [asterisk-users] Help with CDR-Stats

Right, thanks for your reply! 2015-12-16 14:45 GMT-02:00, Bruce Ferrell : > billing is sending invoices for calls to customers. > > reporting is overall statistics on the aggregate of your calls... > Average call hold time, common (or uncommon destinations) etc. If you >

[asterisk-users] Help with CDR-Stats

Hi everyone! I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. Is there others optins for billing? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] Help with CDR-Stats

CDR-STATS is for reporting. A2Billing is for billing... Regards El 16/12/2015 a las 11:15, Vitor Mazuco escribió: Hi everyone! I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. Is there others optins for billing? Thanks --

Re: [asterisk-users] Help with CDR-Stats

Humm whats is the diferent? Em 16/12/2015 14:19, "Annus Fictus" escreveu: > CDR-STATS is for reporting. > > A2Billing is for billing... > > Regards > > El 16/12/2015 a las 11:15, Vitor Mazuco escribió: > >> Hi everyone! >> >> I'm trying to install CDR-Stats

Re: [asterisk-users] Help with CDR-Stats

billing is sending invoices for calls to customers. reporting is overall statistics on the aggregate of your calls... Average call hold time, common (or uncommon destinations) etc. If you see a destination that suddenly has a lot of calls with hold time below normal, there may be a call

[asterisk-users] Help - Asterisk SIP Messages Parameter Modification

Dear All, I have a query. I want to know if there is any possiblity to modify SIP Messages Parameters using the asterisk CLI mode. I want to change the parameters for e.g in INVITE message. How it can be done in asterisk. Kindly assist me. Regards, *Waleed A. Khan* --

[asterisk-users] Help with voicemail

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec

Re: [asterisk-users] Help with voicemail

Matthew Jordan schrieb: > Do you have a g729 codec module loaded? If so, does it show a Bingo! > translation path between g729 and gsm? If so, do you have sufficient > encoder/decoder licenses? I don't have a translation path between g729 and gsm... Since I don't have a

Re: [asterisk-users] Help with voicemail

On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello wrote: > Hi list! > > My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a > voicemail. > On two of these numbers the voicemail works without any problem, on the other > it doesn't... > I get this

Re: [asterisk-users] Help with voicemail

Check your phone codecs. It set to g729 while you don't have this codec in your asterisk nor files in this codec. בתאריך 17 באוק' 2015 18:34,‏ "Luca Bertoncello" כתב: > Hi list! > > My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a > voicemail. >

Re: [asterisk-users] Help With Physical Layer

-Commercial DiscussionAsunto: Re: [asterisk-users] Help With Physical LayerOn Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi Tony I'm not familiar with the card you but 120 ohm is usually twisted pair, and 75 ohm is coax (usually). If it is changeable its usually done

Re: [asterisk-users] Help With Physical Layer

Dear David, I am sorry, I can give the answer right now as the box is at a remote location where I don't have access right now. However, I think the card is ok and al the spans are working. Yesterday I had asked the telco to bring a new RAD modem and I also took there another dell optiplex 3010

Re: [asterisk-users] Help With Physical Layer

Hi Dale, Yes, I tried a cross-over cable, I also tried terminated a new E1- cable with only PINS 1-2 and 4-5 but still no luck. Everything I tried, I would replicate with the other telco's setup and results would be positive. I have a feeling this new telco brought a new model of a RAD modem

Re: [asterisk-users] Help With Physical Layer

On Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi Tony I'm not familiar with the card you but 120 ohm is usually twisted pair, and 75 ohm is coax (usually). If it is changeable its usually done with jumpers on the card. The new Digium cards have no jumpers

Re: [asterisk-users] Help With Physical Layer

Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco

Re: [asterisk-users] Help With Physical Layer (Tony Kasule)

Tim At first should take a look to cable pinout (RAD documents) as pin 1,2, Transmit (output) and 4, 5 Receive (input) for Digium card you should use a straight cable (try to test with new cable one too). Second check Dahdi configuration parameters, use dahdi commands as; dahdi show

Re: [asterisk-users] Help With Physical Layer

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello users, I have a Digium Te235 and asterisk 13 which have worked well

Re: [asterisk-users] Help With Physical Layer

What response do you get to *CLI pri show spans ? On 30 June 2015 at 09:34, Tony Kasule timotsm...@gmail.com wrote: Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello users, I have a Digium

Re: [asterisk-users] Help With Physical Layer

-- Original Message -- From: Tony Kasule timotsm...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 30/06/2015 8:34:47 p.m. Subject: Re: [asterisk-users] Help With Physical Layer Hello, Anyone to help me with this issue

[asterisk-users] Help With Physical Layer

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

to the channels. From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Alex Villací­s Lasso a_villa...@palosanto.com Sent: 08 April 2015 00:33 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villací­s Lasso a_villa...@palosanto.com: El 07/04/15 a las 17:38, Alex Villací­s

[asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates

Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió: I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this

[asterisk-users] help : annoucement queue

Hi everybody, I've a matter with the queue annoucement with the thereare, because if I put just one member in my configuration (member = SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member = SIP/2098 and member

[asterisk-users] help : annoucement queue

Hi everybody, I've a matter with the queue annoucement with the thereare, because if I put just one member in my configuration (member = SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add one member (eg: member = SIP/2098 and member

[asterisk-users] Help! How to make Asterisk support ICE in public network

Hi friends, I am just starting use asterisk for our VoIP server. It works fine in LAN. But when it is deployed in public network(with a public IP), the SIP clients in different NAT fails to communicate with each other. I have set 'icesupport' to 'yes' in sip.conf and set STURN and TURN server

[asterisk-users] help

Dear all, I was going through sip.conf file and i am not able to understand the working and how to test the functionality of below fields. 1.tcpauthlimit 2.tcpauthtimeout any inputs regarding this will appreciated, thanks in advance Thanks SHIVAKUMAR --

Re: [asterisk-users] help

On Fri, Aug 22, 2014 at 6:48 AM, chandapure shiva chandapure.shiv...@gmail.com wrote: Dear all, I was going through sip.conf file and i am not able to understand the working and how to test the functionality of below fields. 1.tcpauthlimit 2.tcpauthtimeout any inputs

Re: [asterisk-users] Help with a bug

On Wednesday 23 Apr 2014, CDR wrote: Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback

[asterisk-users] Help with a bug (CDR)

I fund the issue and it was in my own code. I apologize. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] Help with a bug

Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that

Re: [asterisk-users] Help with a bug

How many seconds later does the file show up? Can you just throw in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second or two of delay be an issue (or does it still not work)? -Josh On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote: Dear friends I

Re: [asterisk-users] Help with a bug

As a second possible solution, instead of Record, could you use MixMonitor, then run StopMixMonitor and THEN do your Playback? That should definitely make sure the recording file is closed and the file handle released. -Josh On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger

Re: [asterisk-users] Help with a bug

: Wednesday, April 23, 2014 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with a bug As a second possible solution, instead of Record, could you use MixMonitor, then run StopMixMonitor and THEN do your Playback? That should definitely make sure

Re: [asterisk-users] Help with a bug

: asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger Sent: Wednesday, April 23, 2014 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with a bug As a second possible solution, instead of Record, could you use MixMonitor

Re: [asterisk-users] Help with a bug

On Wed, 23 Apr 2014, CDR wrote: The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. A simple test: exten = *,n,record(foo.wav) exten = *,n,playback(foo)

Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

Hello Steve, Have you tried to send the automated call to your dialplan instead of the phone? For example, instead of calling SIP/aastra_phone call Local/aastra_phone@auto-answer-context and tweak auto-answer-context from your dialplan as needed. HTH, Ioan On Tue, Jan 28, 2014 at 6:56 PM,

[asterisk-users] [HELP]: Auto-answering calls placed from call files

Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call

Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

On 28/01/14 16:56, Steve McCann wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call

Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann srmcc...@gmail.com wrote: Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm

Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, January 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [HELP]: Auto-answering calls

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

On 1/9/2014 12:12 PM, Jeremy Kister wrote: On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

: Friday, January 10, 2014 1:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11 Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11 Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile, install, and configure the package. Maybe some

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

Wouldn't it be easier in your case to pay somebody to do the job? I doubt that it would take more than a couple of minutes to compile, install, and configure the package. Maybe some things need to get adjusted as the author has abandoned the project (at least there is no longer a project web

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and could share any tricks they used to get it

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

asterisk-users@lists.digium.com Cc: Sent: Thursday, January 9, 2014 12:12 PM Subject: Re: [asterisk-users] help with Cepstral 6 and Asterisk 11 On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile.  I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I

Re: [asterisk-users] help with Cepstral 6 and Asterisk 11

On 1/9/2014 12:12 PM, Jeremy Kister wrote: On 1/8/2014 9:12 PM, Brandon Coale wrote: However, I am not able to get app_swift to compile. I am running Asterisk 11.6.0 and CentOS 6.4 64-bit. I am wondering if anyone else out there has been able to get app_swift working with Asterisk 11 and

[asterisk-users] help with Cepstral 6 and Asterisk 11

Hello, I recently purchased the Cepstral 6 text-to-speech engine (swift), and am now wondering if I should have bought something else. I would like to use Cepstral text to speech like some people use the Festival() or Flite() applications. For example, when I do a core show application

[asterisk-users] Help - DTMF relay in meetme is not reliable

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other

[asterisk-users] Help with decyphering DND status

Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file:

[asterisk-users] Help me understand these log messages

OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can

Re: [asterisk-users] Help me understand these log messages

... an anonyous (not registerted) sip user from 188.161.238.232 was trying to initiate a call to 9725955 and so on... you could enable sip tracing to get more information. maybe you should change the 'allowguest' option in sip.conf..? regards, yves Am 31.05.2013 23:57, schrieb Chris Gentle:

Re: [asterisk-users] Help me understand these log messages

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Saturday, 1 June 2013 9:57 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help me understand

Re: [asterisk-users] Help me understand these log messages

OK, I understand now. I didn't realize allowguest was on by default. I guess I should read more closely. Thanks! On Fri, May 31, 2013 at 5:15 PM, Yves A. yves...@gmx.de wrote: ... an anonyous (not registerted) sip user from 188.161.238.232 was trying to initiate a call to 9725955 and so

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