similar to: SIP authentication on outgoing call

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

Hi there, I'm having a hard time getting outbound calling to my SIP-->PSTN gateway. I continuasly get the following result in my log files: Jun 5 10:07:50 WARNING[1568]: No such host: t2y Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP' Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time I make the following context in my

Hello, I'm having a problem I can't seen to figure out. In a nut shell, I have asterisk running with 4 accounts configured. All accounts work fine for local calling to each other and voicemail. However, only 1 account can make outgoing calls. All the others fail with the following error. If anyone can shed some light on the possible problem or where to look for more info it

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls

I am trying to make Asterisk communicate with a voice switch which doesn't need (and like) authentication on outgoing SIP calls. I have configured it as follows in my sip.conf:   [myswitch] type=friend host=192.168.1.100 port=5060 context=default canreinvite=no To dial out using this switch (it acts as a PSTN gateway) I use this in extensions.conf:   exten =>

Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx> I have already set localnet and

Hello All, I'm just getting into *, and trying to use a Broadvoice account. It works inbound, but Outbound fails no matter what sip.conf parameters I try. From the recent posts here I think it could be: A bad CVS release - I will try to download and build from a new one Broadvoice not challenging and/or Asterisk not responding with an Authorization: in the INVITE header. I am

> Hello fellow asterisk people! > > I have Asterisk listening on port 5061 and SER on port 5060. > > Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. > > My problems are with SIP. I can make incoming calls from SIP to asterisk > and to any of the other networks, but when I try to make an outgoing call > from Asterisk to SER I see the following in

[wrapping disabled to allow for easier review] Yet another SIP authentication problem. I have SER running, and passing calls to a PRI-enabled Asterisk server from a large range of Media Terminal Adapters, and a few other Asterisk systems set up as "clients". I have this PRI-enabled Asterisk server functioning as a very simple media gateway to hand off my toll-free calls to a PRI -

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: [Twilio] type=auth auth_type=userpass password=mysecret username=myun However, my calls using the trunk are rejected with a 403. Using pjsip logging I notice that the outgoing invite does not have an authentication line. Why is Asterisk not sending

I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?) switch. I'm using a config that looks like: [sip_proxy-out] type=peer authuser=208nnnnnnnn remotesecret=xyzzy qualify=100 host=n.n.n.n call-limit=5 nat=no ; sendrpid=yes insecure=no But the Taqua responds to outbound INVITES with 403 Forbidden (oddly, not 401 or 407). Also, from what I can tell, the

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3. Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound calls fail as though the Asterisk server does not see the extensions representing the FXS ports as

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now, with the same SIP configuration, I cannot establish the peer. I've enclosed a SIP log in the hope that someone can help me analyze this failure. I'd guess the issue is NAT related and wondering if someone can spot a problem in the

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls using the trunk are rejected with a 403. Using pjsip >

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any