[asterisk-users] Asterisk Transfer/call patching support

I'm coming to Asterisk from a traditional PSTN environment, so forgive me if I use the wrong Asterisk/SIP terminology. I need to make a product where calls come in go through various menus and based on various configurations perform attended transfers, blind transfers, and patch callers

[asterisk-users] AMI Originate

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 2

...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 3 miliseconds... 2010/10/1 Danny Nicholas da...@debsinc.com [Dan Cropp] Thank you. I was originally using 30, but had the same problem. Dropped the timeout to 3 thinking I wouldn't have to wait 30 seconds to replicate. Guess that's

[asterisk-users] Transfer cmd via AsyncAGI

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-0002

Re: [asterisk-users] Get Channel Variables in AMI Event NewExten

I believe you will have to monitor for the Newexten event, then send an AMI Getvar command. It doesn't make sense to pass all the possible channel variables along with a Newexten event. There may be a ton of extra variables that someone may not want or need on the AMI. Better to have them ask

Re: [asterisk-users] dial and bridge

You could use AsyncAGI to achieve this. Originate the first call (passing in some unique identifier as a variable), then using AMI you will see the channel data. When you see an Event: AysncAGI for that channel (with that id, you have control of the call). Send a Dial Action telling it to

[asterisk-users] WebRTC and Asterisk 12

Anyone know of a tutorial for configuring WebRTC on Asterisk 12 using PJSIP? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001@192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green

Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP

Crashes at this point - From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Tuesday, March 25, 2014 4:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 12.1.1 - Having trouble

Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP

] On Behalf Of Joshua Colp Sent: Tuesday, March 25, 2014 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 12.1.1 - Having trouble setting up PJSIP Dan Cropp wrote: I am trying to make PJSIP work with my Cisco SPA504G phone. I have

[asterisk-users] Need to spoof the callerid using the AMI Originate

We have several customers we need to place outbound calls for (in a single system). May have to place calls for thousands of different caller ids. Customer signs a contract guaranteeing the caller id they provide is owned by them. I have everything setup for AMI Originate and can place the

[asterisk-users] Originate with Caller ID Name

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk

Re: [asterisk-users] Originate with Caller ID Name

/number of the incoming call and set them as the name/number for the Dial? Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, June 26, 2014 3:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users

Re: [asterisk-users] Need to spoof the callerid using the AMI Originate

wrote: Hi, You can use a Local channel in your originate, and have a piece of local dialplan change that for you. Set(CALLERID(num)=x) On 13 June 2014 15:32, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: We have several customers we need to place outbound calls for (in a single

[asterisk-users] PJSIP configuration question

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any

Re: [asterisk-users] PJSIP configuration question

Subject: Re: [asterisk-users] PJSIP configuration question Kia ora, Dan Cropp wrote: I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send

Re: [asterisk-users] PJSIP configuration question

: true Have a great day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Wednesday, December 10, 2014 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re

[asterisk-users] PJSIP configuration question

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the

Re: [asterisk-users] PJSIP configuration question

] On Behalf Of George Joseph Sent: Wednesday, December 10, 2014 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Not sure why

Re: [asterisk-users] PJSIP configuration question

Thank you Joshua. I will make the modifications this morning and give it a try. Have a great day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, December 10, 2014 7:27 PM

Re: [asterisk-users] PJSIP configuration question

Thanks George. I am NATed. I did not obfuscate the 0.0.19.196. That is really what is showing up. The only portion that I hid is the IP address of my box. Have a great day! Dan On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Thanks George

Re: [asterisk-users] PJSIP configuration question

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, December 11, 2014 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question Thank you Joshua. I will make the modifications

Re: [asterisk-users] PJSIP configuration question

a great day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, December 11, 2014 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP

Re: [asterisk-users] PJSIP configuration question

that field in the ACK response to the OK. (Been a long couple weeks) Have a great day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, December 11, 2014 2:24 PM To: Asterisk Users

Re: [asterisk-users] PJSIP configuration question

I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK SIP --- --- Transmitting SIP request (1004

Re: [asterisk-users] PJSIP configuration question

Ugh. I'm having a bad day. The two traces were swapped. The one on Asterisk 13 is PJSIP. The one on Asterisk 12 is using chan_sip. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday

[asterisk-users] PJSIP configuration question

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/800...@outbound.vitelity.net Exten: createcall

Re: [asterisk-users] PJSIP configuration question

Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. Same problem is happening with both of them. Could this be caused by PJPROJECT 2.3? Anyone have any suggestions for what I can try? My boss is giving me until tomorrow to get the PJSIP support working with Vitelity.

Re: [asterisk-users] PJSIP configuration question

: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0. Same problem is happening with both of them. Could this be caused by PJPROJECT 2.3

Re: [asterisk-users] PJSIP configuration question

, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Hi George, Thank you for looking into this. This is behind a nat… Just to be clear...both the pbx and local endpoints are behind the same NAT? [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol

Re: [asterisk-users] PJSIP configuration question

-users] PJSIP configuration question On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Yes, everything is behind the same NAT. For the application I’m working on, the only endpoint is the endpoint to Vitelity. We use AMI to Originate calls from Asterisk

Re: [asterisk-users] PJSIP configuration question

Thanks George. I will remote into and give this a try. Have a great evening! Dan On Dec 15, 2014, at 7:27 PM, George Joseph george.jos...@fairview5.commailto:george.jos...@fairview5.com wrote: Ok Dan, try this... I was able to get this to work behind a NAT and with ip address

Re: [asterisk-users] PJSIP configuration question

I am not sure if I entered the correct settings for the transport information. For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered. One minor detail, we are using ip authentication. When Vitelity changed my

Re: [asterisk-users] PJSIP configuration question

On Dec 15, 2014, at 9:32 PM, George Joseph george.jos...@fairview5.commailto:george.jos...@fairview5.com wrote: On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: I am not sure if I entered the correct settings for the transport information. For the local_net

Re: [asterisk-users] PJSIP configuration question

Thanks George. I will give it a try. Have a great day! Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 11:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [asterisk-users] PJSIP configuration question

: Re: [asterisk-users] PJSIP configuration question On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Thanks George. I will correct my local_net in the morning. Vitelity chan_sip settings I have working, do not have a fromuser. sip.conf settings... I think

Re: [asterisk-users] PJSIP configuration question

Thank you George and Joshua. This can cause major problems. I've rarely (if ever) come across an ALG (that's what that is) that didn't break something. I am contacting the network admin and seeing if he can modify the firewall. I'm a lifelong programmer. Code and programming, I understand.

Re: [asterisk-users] PJSIP configuration question

Here's an update... My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have. He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact

Re: [asterisk-users] originate , callerid

If you are trying to originate a call to the outside PSTN, caller ID could be blocked by your SIP provider. Due to regulations designed to protect consumers, caller ID is generally not trusted in the default configuration. If this is causing your problem, contact your SIP provider and explain

Re: [asterisk-users] PJSIP Endpoint AOR question

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP Endpoint AOR question Dan Cropp wrote: I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set

[asterisk-users] PJSIP Endpoint AOR question

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to

Re: [asterisk-users] PJSIP Endpoint AOR question

, or maybe receive a call from one? On Wed, Apr 1, 2015 at 2:53 PM Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field

[asterisk-users] Transfer

I am running Asterisk 13.5.0. I have the Transfer working when using the chan_sip support. However, when I try to perform a Transfer using pjsip, it is failing. The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By. PJSIP does not. The switch provider

Re: [asterisk-users] pjsip.conf question

Sent: Tuesday, July 14, 2015 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pjsip.conf question Dan Cropp wrote: snip My pjsip.conf looks like... [transport1] type = transport bind = 0.0.0.0 protocol = udp [3400] type = aor

[asterisk-users] pjsip.conf question

[3400] type = aor max_contacts = 1 remove_existing = yes contact=sip:xxx.xxx.xxx.xxx [3400] type = endpoint context = DEF transport = transport1 aors = 3400 accountcode = 1 dtmf_mode = inband device_state_busy_at = 32 Dan Cropp Senior Software Engineer, RD Software Dept. AMTELCO, 4800 Curtin Drive

Re: [asterisk-users] PJSIP add

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, August 25, 2015 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP add Dan Cropp wrote: I am trying to set add a SIP Header to a call

Re: [asterisk-users] PJSIP add

Sorry, replied to the wrong message. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Tuesday, August 25, 2015 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re

[asterisk-users] PJSIP add

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

in the queue on behalf of the agent (replace number with the agent's extension number) In dialplan [agent], wild card match the number, add the header, and then Dial(PJSIP/{$EXTEN}) to send the call to the agent. On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: I

[asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send the call to a macro which does...

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

, read that book, and if you get stuck ask for help. On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp d...@amtelco.commailto:d...@amtelco.com wrote: Thanks Scott. I’m taking over for someone else’s code, so I must admit I’m still learning the Agent and Queue concepts. Local channels are something I

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Den 2015-08-28 kl. 00:07, skrev Dan Cropp: I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include

Re: [asterisk-users] PJSIP add

Of Joshua Colp Sent: Tuesday, August 25, 2015 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP add Dan Cropp wrote: I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Friday, August 28, 2015 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Thank you Niklas

Re: [asterisk-users] PJSIP configuration question

Thank you Joshua. I tried setting the from_domain for the endpoint, but it still sends the internal ip address for the INVITE's From field [acl1] type = acl deny = 0.0.0.0/0.0.0.0 permit = variousaddress permit = bluipaddress [transport1] type = transport bind = 0.0.0.0 protocol = udp

Re: [asterisk-users] PJSIP configuration question

Sent: Tuesday, December 15, 2015 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question Dan Cropp wrote: > outbound_proxy = chi-sbc3-iad.bluip.com Try setting this to: outbound_proxy = chi-sbc3-iad.bluip.com\;lr -- Jos

[asterisk-users] PJSIP configuration question

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my

[asterisk-users] Is it possible to change the default format for ConfBridge recordings?

We have a customer who does significant ConfBridge recording every day. They are concerned about the size of the recording that will accumulate. >From the confbridge.conf.sample file, it mentions "the default format is 8khz >slinear" It is possible to change that "default format" and if so,

[asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

I place a call into Asterisk (from SIP phone) and the To header does not have a tag. Asterisk then sends it's Trying response, still no tag in the To header. The phone then replies with OK, this time the To header includes a tag. Is there any way to retrieve this response To header (including

Re: [asterisk-users] Asterisk 14.2.1 PJSIP - is it possible to retrieve a PJSIP header To field for the SIP OK response to Trying?

:25 PM, Dan Cropp wrote: > I place a call into Asterisk (from SIP phone) and the To header does > not have a tag. Asterisk then sends it's Trying response, still no > tag in the To header. The phone then replies with OK, this time the > To header includes a tag. > >

[asterisk-users] Attended Transfer using AMI on PJSIP

I need to make attended transfer work via an AMI request. Based on data from a Cisco trace from another system which successfully does an attended transfer, the Refer-To header requires the following format Using the

[asterisk-users] Looking for Speech Recognition (ASR) suggestions

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound

Re: [asterisk-users] Looking for Speech Recognition (ASR) suggestions

? Best regards, Luca On Wed, Feb 22, 2017 at 4:43 PM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability fo

Re: [asterisk-users] Looking for Speech Recognition (ASR) suggestions

at 6:35 PM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Hello Luca, Thank you for your response. I’m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR u

Re: [asterisk-users] PJSIP and P-Asserted-Identity

] PJSIP and P-Asserted-Identity Dan Cropp wrote: > > If no caller id is present, calls go through IPitimi to my cell phone. > However, if caller id is present, the P-Asserted-Identity is the > caller id. Based on conversations with IPitimi and some other SIP > products, this is i

[asterisk-users] PJSIP and P-Asserted-Identity

I am working with a customer and their SIP provider is IPitimi. The customer needs to sometimes provide various caller id number for the calls going to IPitimi. They are processing calls for multiple businesses who want their caller id to show up. When no caller id is provided, the From must

[asterisk-users] AMI Originate and 183 response

I to Originate channels using AMI. When the other end indicates the channel is ringing, I need to do some system notification work. Everything works great when the ITSP sends a 180 Ringing response. Through AMI events I see the channel state changed and can do the necessary work. However,

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

(this will not totally disable selinux, but switch it to a permissive mode). For your second point, maybe you should check that firewalld is either stopped or configured properly. By default it has quite restrictive rules. Best regards Jean Aunis Le 14/03/2017 à 17:45, Dan Cropp a écrit : Some

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

to research it to learn what I need to do. Dan Cropp Senior Software Engineer, R Software Dept. AMTELCO, 4800 Curtin Drive, McFarland, WI 53558-9424 608 838-4197 ext. 291 1-800-238-5275 ext 291 www.amtelco.com Statement of Confidentiality The contents of this e-mail message and any attachments

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

Thanks Jason. I will try to explain what I'm seeing for this issue. I did a fresh install of CentOS 7 Minimal into a VM with VMWare Workstation. Followed the Asterisk from Source instructions using pjproject 2.6 and asterisk 13.14.0 for the configure, install, ... At the end of the asterisk

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

Here is the audit.log. Does this indicate a problem with accessing the astdb.sqlite3 file? Permissions for this file are... [root@localhost ~]# ls -l /var/lib/asterisk/astdb.sqlite3 -rw-r--r--. 1 root root 5120 Mar 15 09:39 /var/lib/asterisk/astdb.sqlite3 [root@localhost ~]# tail -f

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

box. I find that this is a handy web/graphical interface to Centos7. On 15/03/2017 10:55 AM, Dan Cropp wrote: Here is the audit.log. Does this indicate a problem with accessing the astdb.sqlite3 file? Permissions for this file are... [root@localhost ~]# ls -l /var/lib/asterisk/astdb.sqlite3

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

s-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Wednesday, March 15, 2017 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] Having problem getting Asterisk to work on

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Wednesday, March 15, 2017 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

[asterisk-users] Having problem getting Asterisk to work on CentOS 7

Some background information. I have used Debian with Asterisk for several years. Have encountered zero problems. I am now trying to setup an Asterisk on a CentOS7 box using VMWare Workstation. I am brand new to CentOS and RHEL so I may be missing something obvious. I am installing CentOS

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

_sigaction(SIGINT, {SIG_IGN, [], SA_RESTORER, 0x7f1a0245b250}, {0x43e670, [], SA_RESTORER, 0x7f1a0245b250}, 8) = 0 wait4(-1, Have a great day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen S

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

to register to that system? On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing

[asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a

[asterisk-users] Asterisk BackgroundDetect and the talk extension

I am using AMI to issue a BackgroundDetect on a channel. Everything works great, I receive the result and the variables on the channel. I am running into one issue though. After calling that function on AMI, when I send the next command on AMI for that channel. For example, a Playback. This

Re: [asterisk-users] Simple speech recognition for driving IVR - "press or say one".

UniMRCP with one of the various speech recognition providers they support definitely works for this. Specify multiple grammars in the MRCP call. One for text to listen for. Another for the DTMFs to listen for. The results will indicate which grammar and what was detected. The combination of

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system? On Thu, Jan 4, 2018 at 11:07 AM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Thank you George. I will pass along the rfc informat

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system? On Thu, Jan 4, 2018 at 11:07 AM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Thank you George. I will pass along the rfc

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

system? On Mon, Dec 18, 2017 at 9:04 AM, Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: Thanks George I originally didn’t have the 1002@ for the identify. Changed that when things were not working. I changed it back. Unfortunately, the system I am connecting

Re: [asterisk-users] How do I retrieve the Call-ID from the SIP INVITE when using Originate on PJSIP?

Please Disregard. I found the solution. In case anyone else runs into this, use (CHANNEL(pjsip,call-id)) and not the PJSIP_HEADER(read, ...) From: asterisk-users On Behalf Of Dan Cropp Sent: Thursday, August 23, 2018 9:32 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How

[asterisk-users] How do I retrieve the Call-ID from the SIP INVITE when using Originate on PJSIP?

Using chan_sip, we are able to retrieve the Call-ID using the SIPCALLID channel variable. This works for both inbound and outbound (Originated) calls. For PJSIP this variable doesn't get set. Instead, for inbound calls we retrieve the Call-ID from the INVITE packet using the

Re: [asterisk-users] PJSIP Originate

14, 2018, at 12:58 PM, Dan Cropp wrote: > I am using AMI Originate to perform a new outbound call. > > The SIP Provider we send the call to wants us to pass the caller id of > the person we are calling for in the Contact header. > > For the AMI Originate, I pass the caller i

[asterisk-users] PJSIP Originate

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being

Re: [asterisk-users] AMI potential memory leak which may be causing a crash

Unfortunately, upgraded to Asterisk 13.20.0 and we are still seeing strange results in the AMI AsyncAGIExec Result string. First one for the call is successful. Later during the same call, it has characters that would be from some ExternalIVR work for this call. The command initiated was an

[asterisk-users] AMI potential memory leak

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following

Re: [asterisk-users] AMI potential memory leak

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, March 22, 2018 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak HI Matt, I am trying to replicate this particular problem. We

Re: [asterisk-users] AMI potential memory leak

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp <d...@amtelco.com> wrote: > We are communicating with Asterisk via AMI. Running Asterisk version > 13.18.5 on an Ubuntu box. > > > > If you look at the event response, the Result field is filled with > random characters. I’m

Re: [asterisk-users] AMI potential memory leak

Of Dan Cropp Sent: Thursday, March 22, 2018 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak We just received a separate call with a Result that seems random... This is on a separate box running Asterisk 14.7.5

Re: [asterisk-users] AMI potential memory leak

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Thursday, March 22, 2018 9:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AMI potential memory leak Not sure if this may

Re: [asterisk-users] After updating to 16 "Some non-required modules failed to load"

The res_pjsip_transport_websocket failing to load seems to be a conflict with the chan_sip.so loading. When I make the chan_sip.so not load, res_pjsip_transport_websocket.so does load. We have customers who need chan_sip and chan_pjsip, so we need to load both. Is there a way to make the

Re: [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"

to g729 stream if the codec negotiation is clearly ulaw in the SIP packets? Dan Cropp Senior Software Engineer, R Software Dept. AMTELCO, 4800 Curtin Drive, McFarland, WI 53558-9424 608 838-4197 ext. 291 1-800-238-5275 ext 291 www.amtelco.com<http://www.amtelco.com/> Statement of Confident

[asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and

Re: [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match"

: [asterisk-users] Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-01d2' when we're sending 'ulaw', switching to match" On Wed, Oct 3, 2018 at 2:09 PM Dan Cropp mailto:d...@amtelco.com>> wrote: I’m reaching out to the asterisk users e-mail list

[asterisk-users] WebRTC using SIPML5 question

I had SIPML5 working with my Asterisk 16 last week. Not sure what I changed, but I'm now receiving the following in asterisk whenever I try to login. Can anyone provide some guidance on what I should be looking at or how to diagnose the problem? [12/12 08:46:18.161] DEBUG[7322] http.c: HTTP

Re: [asterisk-users] WebRTC using SIPML5 question

I figured out my problem. Cleared all browser settings at one point. Once I visited the secure IP address 8089/ws web page directly (accepting unsafe browsing) I was able to login. From: asterisk-users On Behalf Of Dan Cropp Sent: Wednesday, December 12, 2018 10:52 AM To: asterisk-users

[asterisk-users] Question on WebRTC configuration

In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients... https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients "To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows:

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