similar to: PJSIP ports, multiple IP addresses and wrong owner

On Sun, Dec 21, 2014 at 4:54 AM, Recursive <lists at binarus.de> wrote: > Dear list, > > I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. > > 1) Ports and IP addresses which PJSIP bind to > > I have configured one transport

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 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 called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

Dear all, I have the following situation: Local T.38 endpoint <-> ASTERISK <-> SIP provider (with T.38 support) I am trying to send a fax from my local T.38 endpoint to arbitrary external fax numbers (which I am not in control of, so I don't know if the other end supports T.38, is connected to a PBX, who is their provider, and so on), of course trying to use T.38 at least from

I have servers setup in versions 11 and 13. Between two 11 servers, I had no issues sharing BLF, and assigning the hints on my Cisco 525G2 phones. I've upgraded to 13 on one of these servers, and now can't share BLF. I get something like... [2017-07-15 22:35:49] NOTICE[3483]: res_pjsip/pjsip_distributor.c:347 log_unidentified_request: Request from '"Travis Ryan" <sip:612

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the

I've seen this happen three times in the wild now.  I've been trying to isolate the source of the issue, but so far it seems like there's not enough debug output to know why this occurs. Long story short: - Start Asterisk - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind NAT).  SIP is handled correctly, Asterisk responds OK with RTP media address of

On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote: > you can try line functionality on the outbound registration which > may or may not work[2] (requires the upstream to adhere to the RFC, > which not all do). My provider seems to implement this. However even with the line=... in the: SIP to address: sip:5555551212@<my_IP_address>:5060;line=dpnlyiu res_pjsip is still

On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski <markm-lists at intellasoft.net> wrote: > I've seen this happen three times in the wild now. I've been trying to > isolate the source of the issue, but so far it seems like there's not > enough debug output to know why this occurs. > > Long story short: > - Start Asterisk > - PJSIP Handles receiving INVITE from

I'm being told by my ITSP that my Asterisk shouldn't be challenging their system to authenticate (i.e. a 401 response) when they send me a SIP MESSAGE (or I suppose a SIP INVITE for that matter). But I'm not sure what a pjsip.conf configuration for that looks like. How does one associate an incoming call/message with an existing authenticated outgoing registration so that Asterisk

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

Dear list, happy new year! I am still trying to make T.38 work. In the meantime, I have upgraded to Asterisk 13.1.0, and I am using the most recent PJSIP library (compiling that stuff myself). My local fax software is capable of T.38, as is my ITSP; Asterisk sits in the middle, of course. Asterisk is in the same private subnet as the local fax software and talks to the ITSP through a NAT'd

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

Hi guys I have a config as follows for one of my networks. I want to give the xxx.xxx.xxx.xxx/xx network 64kbit for everything from the internet but 8000kbit from our internal servers on yyy.yyy.yyy.yyy/yy network. It does not work. I only want to use u32 filters. I think what's happening is the first flowid of 1:21 is catching them and not getting to the 1:40 flowid. Is this right? The box

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

Hello, I'm helping a colleague (*) which has the following setup: ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>-- Audiocodes MP-112 --- <FXO/FXS> --- Fax machine My issue is the following : Audiocodes gateway reject INVITEs with 488 Not Acceptable Here It seems this gateway requires t38 settings to be present in SDP body in the very first INVITE. My

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

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 the tag field) from the dial plan? I have tried the PJSIP-HEADER read of the To header, but it

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things