similar to: Question on pjsip.conf and aors

On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp <dan at amtelco.com> wrote: > I have the following configuration… > > > > [aor3] > > type = aor > > max_contacts = 1 > > remove_existing = yes > > > > [auth3] > > type = auth > > username = 1004 > > password = SuperSecretProbation > > > > [1004] > > type = endpoint >

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 match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

I just realized that you are asking about dynamic AORs, not static Contacts in an AOR. That may be the difference. I have never actually tried giving a dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, you'd just use an AOR that allowed more than 1 contact. On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard <kctrey at gmail.com> wrote: > I don't know

I don't know why you have issues using different names. I have multiple AORs assigned to a single endpoint and it works fine. I have to admit that my AORs do contain the endpoint name, though. For example, for endpoint "myswitch" I have two AORs, "myswitch_1" and "myswitch_2", and I assign them to the endpoint with aors=myswitch_1,myswitch_2. When you say that

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 match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

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 with doesn?t seem to support the line support. Looking at the SIP packets, I see Asterisk send it. Unfortunately, they do not send the line information as part of the INVITE. I checked with some developers of that system

Thank you George. I will pass along the rfc information to those responsible for the other switch. I missed the match_header addition to Asterisk. Unfortunately, the only header field that seems appropriate is the To header. On a separate box I am now trying to configure the endpoint recognition. Planning on multiple endpoints to the same switch, so I am trying to use the match_header field.

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 allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

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 Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

I am currently running Asterisk 13.1.0-1 I have a chan_sip configuration that works fine with a 3rd party. Third party does not use authentication or registration, it's ip based authentication... When I try switching to PJSIP.conf, I seeing 488 responses from the Asterisk side. What has me really baffled is the debugging indicates [Jul 14 17:28:24] DEBUG[3620] pjsip: sip_endpoint.c

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the

Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I think that means two 'endpoints' in pjsip right? But what exactly is the difference between

Thanks for responding, On Sun, Jan 4, 2015 at 5:45 PM, George Joseph <george.joseph at fairview5.com> wrote: > On Sun, Jan 4, 2015 at 3:29 PM, Antonio G?mez Soto < > antonio.gomez.soto at gmail.com> wrote: > >> Hello, >> >> I am slightly confused by the difference between chan_sip and pjsip. >> Especially the new (to me) objects aor and contact.

Hi, I realise this is an old question, but I’m struggling to get my head around it. The ERD suggests that endpoints can link to multiple AoRs In what situation would you actually use this? Given that mapping of inbound calls is primary done to the endpoint, it looks to me like most of the scenarios where this might be beneficial are actually not possible? One example I had envisaged was being

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*

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 [BLUIPIN] type = aor remove_existing = yes contact = sip:bluipaddress [auth7] type = auth username =

Hi, how come that exim can not authenticate users using PAM? ====================== PLAIN: driver = plaintext server_set_id = $auth2 server_prompts = : server_condition = ${if pam{$auth2:$auth3}{yes}{no}} server_advertise_condition = ${if def:tls_cipher } server_debug_print = "auth: $1 $2 $3" ======================

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 to outbound.vitelity.net At this point, it seems to be working (and this is going through a Cisco

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 sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp <dan at 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? > > > > Anyone have any suggestions for what I can try? > > > > My boss is giving me until