similar to: Load balancing SIP trunks?

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition?

Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did

I'm trying to move my Asterisk deployments under a Virtual IP address and now remember why I dislike this. My primary Asterisk system is now behind a firewall in private address space. My question is what ports are needed to be opened just for the purpose of placing outgoing calls. I would have assumed none, but I can't even get replies on registration from any of my 3 VoIP providers.

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

Hello list I am having an issue with Chanspy/SIP that I?m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: >> but as soon as I configure another sip registration on another server, >> outgoing >> calls drop after 32 seconds. > Are both your servers behind the same NAT router? > thanks for taking part... I don?t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

Hello Asterisk can send calls, but I don't get a call. What could be the problem? [from-siptrunk] exten => 13XXXXXXX,1,dial(${OPERATOR},20) Thanks.

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

I've got a very nice PHP AGI script but I want to be able to do some database cleanup when the user hangs up the phone. I wish everyone would hang up when they were suposed to, but some people don't. So what does Asterisk send to an AGI file when the line has been disconnected? If I remember reading somewhere correctly, I don't need to use DeadAGI. Instead I'm able to use

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

I've heard about this, but I really can't seem to find anything on it. I've got a strange setup that exists only because of firewall issues, and everything about it seems fine. The setup: SIP clients -> Asterisk (office) -> IAX -> Asterisk (colocation) -> SIP PSTN Termination All the extensions I want to be able to dial are on the colocation box. What I'd really