[asterisk-users] sip trunk, parsing DID

Hello, I am using a Swiss VoIP provider called sipcall. They have what they call a SIP trunk, and it is less expensive than individual accounts. From Asterisk's point of view, this is just a regular SIP account, which can however receive and send calls from multiple numbers. I just migrated from i

Re: [asterisk-users] SIP INFO messages with Content-Type: application/media_control+xml

On Wed, Oct 13, 2021 at 10:05 AM Floimair Florian wrote: > Hi all! > > > > We have a WebRTC user-agent (using sip.js) that is giving me headache. > > > > When a different user-agent calls this user-agent, we frequently see > Asterisk generating SIP INFO messages with > > > > Content-Type: applica

[asterisk-users] SIP INFO messages with Content-Type: application/media_control+xml

Hi all! We have a WebRTC user-agent (using sip.js) that is giving me headache. When a different user-agent calls this user-agent, we frequently see Asterisk generating SIP INFO messages with Content-Type: application/media_control+xml Content-Length: 178 In the payload, tha

Re: [asterisk-users] SIP Source Port

Maybe it could be accomplished in the firewall? Tell the firewall to NAT the source port of packets to 5061? Från: asterisk-users-boun...@lists.digium.com För Alexander Perkins Skickat: den 10 juli 2021 19:39 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] SIP Source Port Hi

Re: [asterisk-users] SIP Source Port

El Sat, 10 Jul 2021 23:02:10 +0200 Antony Stone escribió: > On Saturday 10 July 2021 at 22:57:09, Eric Wieling wrote: > > > > On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins wrote: > > > > > > Hi All. We have a provider that requires us to SOURCE the SIP > > > connection on TCP 5061.

Re: [asterisk-users] SIP Source Port

On Saturday 10 July 2021 at 22:57:09, Eric Wieling wrote: > > On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins wrote: > > > > Hi All. We have a provider that requires us to SOURCE the SIP > > connection on TCP 5061. I honestly have no clue how to force > > Asterisk to always SOURCE

Re: [asterisk-users] SIP Source Port

Kamailio is useful when you want to do weird, non-standard, or unusual stuff with SIP. You could send your outgoing connections to Kamailio, which could then send the connection out with the required source port. Have you considered using a not stupid provider? On 7/10/21 3:44 PM, Joshua C.

Re: [asterisk-users] SIP Source Port

On Sat, Jul 10, 2021 at 2:39 PM Alexander Perkins < alexanderhenryperk...@gmail.com> wrote: > Hi All. We have a provider that requires us to SOURCE the SIP connection > on TCP 5061. I honestly have no clue how to force Asterisk to always > SOURCE the SIP connection on a certain port. > > Can any

Re: [asterisk-users] SIP Source Port

] On Behalf Of Alexander Perkins Sent: Saturday, July 10, 2021 1:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Source Port Hi All. We have a provider that requires us to SOURCE the SIP connection on TCP 5061. I honestly have no clue how to force Asterisk to always

[asterisk-users] SIP Source Port

Hi All. We have a provider that requires us to SOURCE the SIP connection on TCP 5061. I honestly have no clue how to force Asterisk to always SOURCE the SIP connection on a certain port. Can anybody point me in the right direction? I am using PJSIP. Thank you, Alex --

Re: [asterisk-users] SIP Realtime peers

On Thursday 08 April 2021 at 12:38:02, Antony Stone wrote: > On Thursday 11 March 2021 at 14:03:23, Antony Stone wrote: > > Hi. > > > > I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 > > and 16. > > > > In general it's all working fine, however there's something that p

Re: [asterisk-users] SIP Realtime peers

On Thursday 11 March 2021 at 14:03:23, Antony Stone wrote: > Hi. > > I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 > and 16. > > In general it's all working fine, however there's something that puzzles > me: > > If I connect to the Asterisk console and use the comman

Re: [asterisk-users] SIP Realtime peers

Hi. Has nobody else tried to do this, or worked out how to (or, possibly, reported it as a bug)? On Saturday 13 March 2021 at 16:13:04, Antony Stone wrote: > On Thursday 11 March 2021 at 14:03:23, Antony Stone wrote: > > Hi. > > > > I'm using MariaDB (via ODBC) to store realtime SIP peers with

Re: [asterisk-users] SIP Realtime peers

On Thursday 11 March 2021 at 14:03:23, Antony Stone wrote: > Hi. > > I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 > and 16. > > In general it's all working fine, however there's something that puzzles > me: > > If I connect to the Asterisk console and use the comman

[asterisk-users] SIP Realtime peers

Hi. I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 and 16. In general it's all working fine, however there's something that puzzles me: If I connect to the Asterisk console and use the command "sip show peers" I get a list of the peers including the last qualify time

[asterisk-users] SIP TLS, Not HTTPS

Hi All. We are trying to get SIP TLS working, but have run into a snag. We followed this documentation - https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial - but, when it comes to PJSIP, we are bit lost on the authentication process. For example, in that documentation, the peer ha

[asterisk-users] SIP/2.0 401 Unauthorized

Hello I use Asterisk 13 with FreePBX. When I try to connect my Softphone via VPN to Asterisk I'm registered and It's show via "pjsip list contacts" Then I try to call an internal number / other extension I get the following: "SIP/2.0 401 Unauthorized". The VPN net is list in pjsip.transports.conf

Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0

Hi Stefan, thanks a lot. It is working now. Best regards, Karsten Am Freitag, den 01.05.2020, 18:40 +0200 schrieb Stefan Tichy: > Hi Karsten, > > > On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > > > > The server sends Server Hello, Certificate, Server Key > > Ex

Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0

Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster.

[asterisk-users] SIP TLS not working, Asterisk 16.9.0

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Descr

Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

On 23/03/2020 11:29, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 7:15 AM John Hughes > wrote: Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presenc

Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

Because Asterisk do not support PUBLISH. For BLF Configuration: https://wiki.asterisk.org/wiki/display/AST/Configuring+chan_sip+for+Presence+Subscriptions or https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+for+Presence+Subscriptions --- I'm SoCIaL, MayBe On 3/23/20 05:13, Jo

Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

On Mon, Mar 23, 2020 at 7:15 AM John Hughes wrote: > Hi, in these dark days of COVID-19 lockdown I'm using linphone to > connect to my office asterisk system for working from home. > > It's going pretty well but the presence/BLF functions don't appear to work. > > In the linphone logs and asteris

[asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH

Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

"Joshua C. Colp" writes: >> I am curious if the "reuse registration TCP connection" is required by >> standards or if it is merely obviously good practice. >> >> I have had this problem too with asterisk 16.5.0 >> >> This is not the first recommendation I have seen to use kamailio as a >> proxy f

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

On Fri, Dec 27, 2019 at 2:00 PM Greg Troxel wrote: > Dovid Bender writes: > > > So long as the tcp socket is open your SBC should send the call back over > > the same socket. Now it can be that your SBC is seeing the socket as > > timing out. If you are using Kamailio you can have it send tcp ke

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

Dovid Bender writes: > So long as the tcp socket is open your SBC should send the call back over > the same socket. Now it can be that your SBC is seeing the socket as > timing out. If you are using Kamailio you can have it send tcp keep alives > every so often so that the socket stays up. SBC?

Re: [asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

So long as the tcp socket is open your SBC should send the call back over the same socket. Now it can be that your SBC is seeing the socket as timing out. If you are using Kamailio you can have it send tcp keep alives every so often so that the socket stays up. On Fri, Dec 27, 2019 at 10:41 AM B

[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

Hi List I wonder how SIP via TCP is supposed to work. Not realy Asterisk related, but I hope you experts might be able to help out :-) One of our customers has a SIP device registering via a complex NAT. To benefit from TCP Connection Tracking, he choose TCP instead of UDP. So he expected, that

[asterisk-users] SIP messaging

how would one set up SIP messaging from one server to another server and based on the reply back from the server - perhaps do something else . I think I can do receiving the SIP message - but how about the response back. I was looking around and have not found an example. Thanks, Jerry -- _

Re: [asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

On Wed, Sep 4, 2019, at 6:01 AM, bilal ghayyad wrote: > Hello; > > I am facing a trouble with the SIP service provider, they are saying > that there is a problem related to message option 200 (the heartbeat), > so what is required to add this for the sip configuration? Below is my > sip debug t

[asterisk-users] SIP trunk problem: Message option 200 (heartbeat)

Hello; I am facing a trouble with the SIP service provider, they are saying that there is a problem related to message option 200 (the heartbeat), so what is required to add this for the sip configuration? Below is my sip debug trace log with the them and the sip peer configuration: [Sep  4 12

Re: [asterisk-users] SIP trunk between asterisk boxes

On Tue, Jul 23, 2019, at 2:53 PM, Jerry Geis wrote: > > rtp set debug on" will show the RTP traffic flowing,I did not see anything > > printed when I pressed a key. I say the audio prints. That means either it was not negotiated or was not picked up by Asterisk. -- Joshua C. Colp Digium - A Sa

Re: [asterisk-users] SIP trunk between asterisk boxes

> rtp set debug on" will show the RTP traffic flowing, I did not see anything printed when I pressed a key. I say the audio prints. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out th

Re: [asterisk-users] SIP trunk between asterisk boxes

On Tue, Jul 23, 2019, at 1:47 PM, Jerry Geis wrote: > I have a sip trunk between two asterisk boxes. > I can call into the first box, hit 499 for example and the call goes to > the second box and answers as expected plays me audio message just fine > etc... My issue is that DTMF does not seem to

[asterisk-users] SIP trunk between asterisk boxes

I have a sip trunk between two asterisk boxes. I can call into the first box, hit 499 for example and the call goes to the second box and answers as expected plays me audio message just fine etc... My issue is that DTMF does not seem to be working. Both sides are set for: dtmfmode=RFC2833 What mi

Re: [asterisk-users] SIP credentials in the dialplan

Josh, Thanks. I had another look. This seems to work for me: Dial(SIP/18005551212:PASSWORD::usern...@sip1.mydomain.net!! usern...@sip1.example.net,,) So it seems like I needed to put the called number followed by the password :: and then the username. On Tue, Jul 9, 2019 at 8:57 AM Joshua C. Co

Re: [asterisk-users] SIP credentials in the dialplan

On Tue, Jul 9, 2019, at 9:46 AM, Dovid Bender wrote: > > > On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp wrote: > > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > > Hi, > > > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > > should be able to dia

Re: [asterisk-users] SIP credentials in the dialplan

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent versi

Re: [asterisk-users] SIP credentials in the dialplan

On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > Hi, > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > should be able to dial with SIP credentials in the DP. Is this still > possible in recent versions of Asterisk either with chan_sip or pj_sip? PJSIP does not

[asterisk-users] SIP credentials in the dialplan

Hi, Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you should be able to dial with SIP credentials in the DP. Is this still possible in recent versions of Asterisk either with chan_sip or pj_sip? TIA. Dovid --

[asterisk-users] SIP reinvite with ice restart

Hello, Does Asterisk PJSIP support an ice restart from UA ? When i make some tests, Asterisk reply SDP without ICE candidate and the setRemoteDescription on my UA failed because there is no ice candidate. I read the RFC https://tools.ietf.org/html/rfc8445#section-9 and it not clear for me, i

Re: [asterisk-users] SIP Codec negotiation

On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote: So, Asterisk will defer it's choice of codec to match the codec it detects in the incoming stream? On Fri, 11 May 2018, Joshua Colp wrote: It depends on the channel driver and configuration. The chan_sip module always matching outgoing

Re: [asterisk-users] SIP Codec negotiation

The unwritten rule of SDP is that if possible you use the first codec of a type listed, but you don’t have to. If the sender says he can do something, he had better be prepared to handle media of that type no matter in what order it was listed. So when you send OK with ulaw as first priority and

Re: [asterisk-users] SIP Codec negotiation

On Fri, May 11, 2018, at 10:36 AM, Steve Edwards wrote: > On Fri, 11 May 2018, Joshua Colp wrote: > > >> In the above example, even though the INVITE/SDP says they prefer gsm > >> over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose > >> to use gsm or ulaw? > > > > Yes. > > > >>

Re: [asterisk-users] SIP Codec negotiation

On Fri, 11 May 2018, Joshua Colp wrote: In the above example, even though the INVITE/SDP says they prefer gsm over ulaw and the OK/SDP says I prefer ulaw over gsm, they can choose to use gsm or ulaw? Yes. Can it be asymmetrical? They send gsm and I send ulaw? Technically, yes. In practice

Re: [asterisk-users] SIP Codec negotiation

On Fri, May 11, 2018, at 10:07 AM, Steve Edwards wrote: > > So, without examining the RTP, you cannot tell which codec was actually > used? >From an Asterisk perspective "core show channel" will also show you what is >currently flowing. > In the above example, even though the INVITE/SDP say

Re: [asterisk-users] SIP Codec negotiation

On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc

Re: [asterisk-users] SIP Codec negotiation

On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: > I receive an INVITE/SDP containing: > > m=audio 11310 RTP/AVP 3 0 101 > > which I interpret as gsm, ulaw, rfc2833. > > and I reply with an OK/SDP containing: > > m=audio 15884 RTP/AVP 0 3 101 > > which I interpret as

[asterisk-users] SIP Codec negotiation

I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call?

Re: [asterisk-users] Sip cause and response codes in dialplan

: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Sip cause and response codes in dialplan On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote: > Hi, > > I am experimenting with getting hold of the sip cause and sip response > from o

Re: [asterisk-users] Sip cause and response codes in dialplan

On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote: > Hi, > > I am experimenting with getting hold of the sip cause and sip response from > outgoing call. How could i make a userevent printing the sip cause and/or > sip response. I have tried using hangupcause, sip_cause and such , bu

[asterisk-users] Sip cause and response codes in dialplan

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 rea

Re: [asterisk-users] sip trunk with social media

On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote: > Hello > It will be amazing if possible to do sip trunk with any of social media > providers like: whatsapp, facebook, imo, viber, ... etc To the best of my knowledge none of the services you mention either operate over SIP or provid

[asterisk-users] sip trunk with social media

Hello It will be amazing if possible to do sip trunk with any of social media providers like: whatsapp, facebook, imo, viber, ... etc.Did anyone has luck with this? RegardsBilal Sent from Yahoo Mail on Android-- _ -- Bandwidth

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

> fail2ban is most useful for blocking registration attempts.    I > handle  > non-registration call attempts by allowing guests, point them to a > jail  > context, which runs Log(WARNING,fail2ban='${CHANNEL(peerip)}')   I > set a  > fail2ban rule to match that line logged from Asterisk. Thanks

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

IMHO, manual IP-tables is probably better for those who have a single provider - whitelist only your SIP trunk provider's IP adress (or address pool). But... that leads onto a train of thought that might help. First, realize you don't have to manually read your security logs, you can script that

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 01/02/2018 05:30 PM, sean darcy wrote: On 12/30/2017 08:18 PM, Dovid Bender wrote: Script kiddies trying to find vulnerable systems that they can make calls on. Lock down the box with iptables and use fail2ban to block them. The via is probably bogus unless a box at the DoD was comprimised

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 12/30/2017 08:18 PM, Dovid Bender wrote: Script kiddies trying to find vulnerable systems that they can make calls on. Lock down the box with iptables and use fail2ban to block them. The via is probably bogus unless a box at the DoD was comprimised. On Sat, Dec 30, 2017 at 6:49 PM, sean d

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 12/30/2017 08:10 PM, Antony Stone wrote: On Sunday 31 December 2017 at 00:49:17, sean darcy wrote: I've been getting a lot of timeouts on non-critical invite transactions. So how is someone on a Dutch ISP using my server to mess with a US DoD ip address ? What's your setting for "allowg

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

Script kiddies trying to find vulnerable systems that they can make calls on. Lock down the box with iptables and use fail2ban to block them. The via is probably bogus unless a box at the DoD was comprimised. On Sat, Dec 30, 2017 at 6:49 PM, sean darcy wrote: > I've been getting a lot of timeo

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On Sunday 31 December 2017 at 00:49:17, sean darcy wrote: > I've been getting a lot of timeouts on non-critical invite transactions. > So how is someone on a Dutch ISP using my server to mess with a US DoD > ip address ? What's your setting for "allowguest" (under [general]) in /etc/asterisk/si

[asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4

Re: [asterisk-users] SIP trunks going to the wrong context

On Thursday, December 14, 2017, 10:05:23 PM, Tony Mountifield (t...@softins.co.uk) wrote: > So I think you really do need to have a single peer section for all sipgate > calls, pointing to one sipgate context in your dialplan that contains all > your various extensions like se2489, sj0151, etc. T

Re: [asterisk-users] SIP trunks going to the wrong context

In article <1513290358.2926.4.ca...@linuxista.com>, Frank Vanoni wrote: > I don't know if it applies to your problem, but I also had some > troubles with multiple account on same SIP provider.  > Here what works for me: > > > In sip.conf: > > > register => 11:qwe...@sip.provider.zz/11

Re: [asterisk-users] SIP trunks going to the wrong context

I don't know if it applies to your problem, but I also had some troubles with multiple account on same SIP provider.  Here what works for me: In sip.conf: register => 11:qwe...@sip.provider.zz/11 ; Trunk1 register => 22:asd...@sip.provider.zz/22 ; Trunk2 register => 22:yxc..

Re: [asterisk-users] SIP trunks going to the wrong context

In article , Ade Vickers wrote: > Hi all, > > I'm trying to resolve a weird issue with SIP routing. > > I have a number of SIP trunks, from a selection of providers, all of > which are registered in sip.conf: > > [general] > context=default > allowguest=no > allowoverlap=no >

[asterisk-users] SIP trunks going to the wrong context

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0

Re: [asterisk-users] SIP-Header Pass-Through

My only suggestion would be you could reduce your line count by replacing your GotoIf statements with ExecIF statements. exten => addheader,1,ExecIf($["x${ARG1}" != "x"]?Set(PJSIP_HEADER(add,Route)=${ARG1})) same => n,ExecIf($["x${ARG2}" != "x"]?Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2}))

Re: [asterisk-users] SIP-Header Pass-Through

On Wed, Aug 2, 2017, at 10:39 AM, Carsten Bock wrote: > Hi, > > quick question: > I need to pass-through some headers from the A-Leg to the B-Leg, which > are connected using PJSIP. > Currently I do the following: > > [handler] > exten => addheader,1,GotoIf($["${ARG1}" == ""]?3) > exten => addhea

[asterisk-users] SIP-Header Pass-Through

Hi, quick question: I need to pass-through some headers from the A-Leg to the B-Leg, which are connected using PJSIP. Currently I do the following: [handler] exten => addheader,1,GotoIf($["${ARG1}" == ""]?3) exten => addheader,2,Set(PJSIP_HEADER(add,Route)=${ARG1}) exten => addheader,3,GotoIf($["

Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

> Not sure maybe there's a better solution but I thought about using another > peer with type=user for incoming connections. That's what I've done for my connection to the service provider I use (Vitelity), as they have different inbound and outbound hosts/proxies. This works fine. -- _

Re: [asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

Not sure maybe there's a better solution but I thought about using another peer with type=user for incoming connections. On Mon, May 22, 2017, 6:13 PM Benoit Panizzon wrote: > Hello List > > I work at an SIP Provider and we have added and SBC in front of our > Voice Switch to protect it. > > Thi

[asterisk-users] SIP Trunk over Proxy (matching ip of outbound proxy in incomming calls)

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => usern...@sip.example.com:p

Re: [asterisk-users] SIP and Voice on different nets

Seems I responded the same time as Josh. Follow what he has suggested. On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev wrote: > Yes, Voice = RTP > > Using chan_sip > > 2017-04-27 15:32 GMT+03:00 Dovid Bender : > >> By voice do you mean RTP? Are you using chan_sip or pjsip? >> >> >> On Thu, Apr

Re: [asterisk-users] SIP and Voice on different nets

Yes, Voice = RTP Using chan_sip 2017-04-27 15:32 GMT+03:00 Dovid Bender : > By voice do you mean RTP? Are you using chan_sip or pjsip? > > > On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev > wrote: > >> ​I have connection with two networks (by VoIP provider setup) >> 1 - 10.10.10.0/24 = SIP >>

Re: [asterisk-users] SIP and Voice on different nets

By voice do you mean RTP? Are you using chan_sip or pjsip? On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev wrote: > ​I have connection with two networks (by VoIP provider setup) > 1 - 10.10.10.0/24 = SIP > 2 - 10.10.11.0/24 = Voice > > How to tell Asterisk send / receive voice traffic not on S

Re: [asterisk-users] SIP and Voice on different nets

On Thu, Apr 27, 2017, at 09:10 AM, Artem Chekulaev wrote: > ​I have connection with two networks (by VoIP provider setup) > 1 - 10.10.10.0/24 = SIP > 2 - 10.10.11.0/24 = Voice > > How to tell Asterisk send / receive voice traffic not on SIP network. > When > I look into dumps, I see Asterisk tryin

[asterisk-users] SIP and Voice on different nets

​I have connection with two networks (by VoIP provider setup) 1 - 10.10.10.0/24 = SIP 2 - 10.10.11.0/24 = Voice How to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voice Unfortunately, I _need_ to use two networks

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Hi Ernie, When one-way audio appear (no matters if there is a VPN or NAT server on the diagram) I simply : * Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x' on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want to debug. * Make a test call and replica

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

On 4/18/2017 7:40 PM, Ernie Dunbar wrote: Server network: 192.168.0.0/24 OpenVPN network: 10.8.0.0/24 Asus network: 192.168.1.0/24 The Asterisk SIP registration appears to be responding properly to this - this is what I see when I do a 'sip show peer' for an Aastra phone that's connecting thro

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

rs@lists.digium.com> Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.   Hi everyone. I'm having some trouble with an OpenVPN tunnel that

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

l Discussion'" >> >> Sent: 19-Apr-17 10:25:59 AM >> Subject: [asterisk-users] SIP connections over OpenVPN connection get >> one-way voice. >> >>> Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't >>> working *qu

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

rs@lists.digium.com> Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.   Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we&#

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

te table. Originalmeddelande Från: Ernie Dunbar Datum: 2017-04-19 00:25 (GMT+01:00) Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Rubrik: [asterisk-users] SIP connections over OpenVP

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

zarre routing problems when states appear in the state table. Originalmeddelande Från: Ernie Dunbar Datum: 2017-04-19 00:25 (GMT+01:00) Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Rubrik: [asterisk-users] SIP connections over OpenVPN connection get

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

-- Original Message -- From: "Ernie Dunbar" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice. Hi everyone. I'm having

[asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Hi everyone. I'm having some trouble with an OpenVPN tunnel that isn't working *quite* as well as we'd hoped. First, here's our technical details: The OpenVPN server (v2.3.4-5+deb8u1) is a Debian 8 box behind a NAT router. The router has UDP port 1194 forwarded

Re: [asterisk-users] SIP peer authentication

type=peer matches on the IP of the specified host, If you want to match on the username use type=user. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: http

[asterisk-users] SIP peer authentication

I have two asterisk boxes connected via SIP protocol. I want to deploy SIP peer authentication to this connection. What is the needed configuration?I have the following configuration but changing username and secret does not affect the connection at all! sip.conf in box 68:[general] t3

[asterisk-users] SIP option for Google Fi or Ting.com or similar - 911 is difficult.

​Want to integrate my cellular service into my asterisk dial plan, so this requires SIP capabilities such as one can get with Vitelity vMobile. Does Google Project Fi or Ting or another offer a SIP integration option, but not advertise it? A drawback with Vitelity vMobile is that it will not comp

Re: [asterisk-users] SIP reload not changing codecs

Hello, on Asterisk 13.13.1 working correctly Regards El 27/02/2017 a las 10:59, Steve Edwards escribió: Asterisk 13.3.2 I change the allowed codec from ulaw to g729 in sip.conf and enter 'sip reload' on the console, but calls continue to use ulaw until restart. Before reload: lc10*CLI>

[asterisk-users] SIP reload not changing codecs

Asterisk 13.3.2 I change the allowed codec from ulaw to g729 in sip.conf and enter 'sip reload' on the console, but calls continue to use ulaw until restart. Before reload: lc10*CLI> sip show settings Global Signalling Settings: --- Codecs: (ulaw) lc

Re: [asterisk-users] SIP host name resolution

Hi, Am 03.02.2017 um 18:23 schrieb Steve Edwards: > If I have a SIP endpoint defined in sip.conf using a host name instead of an > IP address, do I have to reload sip to get Asterisk to 're-resolve' the host > name if I change the IP address in my DNS? Normally, Asterisk honours DNS TTL and wi

[asterisk-users] SIP host name resolution

If I have a SIP endpoint defined in sip.conf using a host name instead of an IP address, do I have to reload sip to get Asterisk to 're-resolve' the host name if I change the IP address in my DNS? Does the answer change if the host name in sip.conf resolves to a CNAME and I change the CNAME in

Re: [asterisk-users] sip show [general]?

Hi You can do sip show settings On Jan 11, 2017 5:32 AM, "Thufir Hawat" wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I output those settings, and only those > s

Re: [asterisk-users] sip show [general]?

'sip show settings' may do what you want. On Wed, Jan 11, 2017 at 3:32 AM, Thufir Hawat wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I output those settings, and o

[asterisk-users] sip show [general]?

I appreciate that the console lets you see the details for a peer with "sip show peer foo". Certainly, I can look in sip.conf to see the [general] context, but can I output those settings, and only those settings, to the console? thanks, Thufir -- _

Re: [asterisk-users] SIP and RTP port and IP addresses

On Thu, Nov 10, 2016 at 7:15 AM, Ethy H. Brito wrote: > On Thu, 10 Nov 2016 00:35:54 +0100 > Max Grobecker wrote: > > > Hi Ethy, > > Hi Max and All. > > > > > > > Am 09.11.2016 um 17:13 schrieb Ethy H. Brito: > > > > > How are these parameters available from dialplan? > > > > > > For instance, $

Re: [asterisk-users] SIP and RTP port and IP addresses

On Thu, 10 Nov 2016 00:35:54 +0100 Max Grobecker wrote: > Hi Ethy, Hi Max and All. > > > Am 09.11.2016 um 17:13 schrieb Ethy H. Brito: > > > How are these parameters available from dialplan? > > > > For instance, ${SIPURI} holds the internal "IP:port" if the client is > > behind NAT. I nee

  1   2   3   4   5   6   7   8   9   10   >