[asterisk-users] PJSIP

Hello, Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I got the "from" to be the way I need it. From: I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be To: sip

[asterisk-users] PJSIP question

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip -- _ -- Bandwidth an

[asterisk-users] PJSIP question

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SI

[asterisk-users] PJSIP question

In a PJSIP endpoint, how do I set all no-named settings so they get inherited from another place and I don't need to mention them again and again for all my endpoints? In regular sip you could specify those options and they remained valid if not redefined by a peer. A case would be the codecs allow

[asterisk-users] PJSIP incompatibility

Dear friends After spending few days converting my app to PJSIP, today I had to roll back the upgrade because in the SDP, the Owner section is wrong, or I misconfigured something This is what my client said: "That OK message from 1.1.1.1 can not be parsed by our switch due to address representatio

[asterisk-users] pjsip: outofcall_message_context

Hello. Is there an analog option "outofcall_message_context" for pjsip? or: how to determine that the "call" is an outbound text message? Dmitriy Serov. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] PJSIP CCSS

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www

[asterisk-users] PJSIP add

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 call

[asterisk-users] PJSIP subscribe

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In

[asterisk-users] PJSIP OPTIONS

Hello Everyone, How to configure PJSIP to reply 200 OK from upstream sip proxy on keepalive packet ? proxy ~> Keepalive OPTIONS ~> asterisk <~ 200 OK <~ volga629 -- _ -- Bandwidth and Colocation Provided by http://www.ap

[asterisk-users] PJSIP Originate

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being

[asterisk-users] PJSIP Qualify

I am currently not using qualify, but it seems like a nice way to know if the phones are online. I attempted to set it up, but am running into a 404 on the subscription. 1. From the manager, Action: PJSIPNotify (with an endpoint). This caused the following OPTIONS packet to be sent to the phone

[asterisk-users] PJSIP Qualify

Please ignore my previous email. Qualify is now working well. The packets that were misleading me were related to MWI subscription, and not the result of Qualify (OPTIONS).. Sorry for any confusion. -- _ -- Bandwidth and Colocat

[asterisk-users] PJSIP reInvite

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvit

[asterisk-users] PJSIP crashes

PJISP cannot handle the From field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax error exception when parsing 'From' header on line 4 col 40: CANCE

[asterisk-users] PJSIP crashes

> > I have no control over the SIP calls I receive. PJSIP should log a warting > and continue. It is causing the CPU usage to spike dramatically. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

[asterisk-users] PJSIP Lockup

Hello All, I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But recently upgraded to attempt to resolve this issue. Using bundled PJSIP. The PBX is using mysql realtime for most functions. The Mysql server is on the same lan as the asterisk box. As more users have been moved to this

Re: [asterisk-users] PJSIP

On Sat, May 30, 2020 at 2:02 AM John T. Bittner wrote: > *Hello,* > > > > Anyone know how to set the “To:” in an invite for PJSIP to custom > settings. I got the “from” to be the way I need it. > > > > From: > To: "TEST" > > > > I have tried a lot of changes to get to this but nothing works. >

[asterisk-users] PJSIP question urgent

I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)}, so if I use PJSIP, for scalability, how do I read what the signalling IP where the inbound call is coming from and what is the inbound codec? You would think that the new channel would set those variables up, isn't it? Philip Or

Re: [asterisk-users] PJSIP question

On Mon, Sep 23, 2013 at 5:20 AM, CDR wrote: > I am stuck in channel PJSIP trying to see the real flow of SIP > messages, what in regular sip > we used to type "sip set debug on" > Also, is there an automated way to convert sip.conf options to pjsip.conf? > First, please keep in mind that Asteris

[asterisk-users] PJSIP Identify Wiky

The Wiky needs to be updated https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29 This is the example shown: "[6001] endpoint=6001 match=203.0.113.1" It should be: "[6001] type=identify endpoint=6001 match=203.0.113.

[asterisk-users] PJSIP and ARA

Hi This is a bit of an exploratory question for groundwork before I start playing with asterisk 12. I've spotted the very useful looking file contrib/realtime/mysql/mysql_config.sql in the source. Are the table names starting ps_ all to do with PJSIP? Direct MySQL connection has been deprecate

Re: [asterisk-users] PJSIP question

On Wed, Jun 18, 2014 at 6:05 AM, CDR wrote: > A few months ago I started using and had to abandon PJSIP because my > dialplan could not read the inbound signalling IP address, which I can > read now in Asterisk11 using CHANNEL(recvip). My app relies on this > information. The > question is, is it

Re: [asterisk-users] PJSIP question

You can use templates. Templates are defines by putting a “(!)” next to the context name. [template-name](!) disallow=all allow=ulaw Then define the template next to the endpoint in parenthesis. [endpoint-name](template-name) Chad On Jun 26, 2014, at 12:30 PM, CDR wrote: > In a PJSIP endpoint

[asterisk-users] pjsip phoneprov realtime?

Howdy, Is there a way to use realtime with phoneprov.com and pjsip? I've got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision. I was hoping the Sorcery page in the wiki would help possibly but it's blank :( https

[asterisk-users] PJSIP configuration question

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestio

[asterisk-users] PJSIP configuration question

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

[asterisk-users] PJSIP configuration question

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/800...@outbound.vitelity.net Exten: createcall C

[asterisk-users] PJSIP configuration question

I am running PJPROJECT 2.3 and Asterisk 13.0.0. I answer the call, about 15 seconds later, vitality hangs up on my cell phone. However, Asterisk is never notified When the OK (for the answer) occurs, the ACK seems to never be accepted. The OK recvd with ACK sent occurs several times. Here are the

Re: [asterisk-users] pjsip: outofcall_message_context

On Wed, Mar 18, 2015 at 4:43 AM, Dmitriy Serov wrote: > Hello. > > Is there an analog option "outofcall_message_context" for pjsip? > or: how to determine that the "call" is an outbound text message? > The 'message_context' endpoint option [1] should provide what you're looking for. [1] https:/

Re: [asterisk-users] PJSIP CCSS

Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? If CCSS is needed then the only option is to use chan_sip. The chan_pjsip module do

Re: [asterisk-users] PJSIP CCSS

Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a écrit : If CCSS is needed then the only option is to use chan_sip. The chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" s

Re: [asterisk-users] PJSIP CCSS

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a écrit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so c

Re: [asterisk-users] PJSIP CCSS

Ludovic Gasc wrote: 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard mailto:jd.gir...@sysnux.pf>>: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a écrit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does

Re: [asterisk-users] PJSIP CCSS

2015-05-21 17:59 GMT+02:00 Jean-Denis Girard : > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a écrit : > > If CCSS is needed then the only option is to use chan_sip. The > > chan_pjsip module does not implement CCSS in any way. > > Is CCSS support planned f

Re: [asterisk-users] PJSIP CCSS

2015-05-21 18:43 GMT+02:00 Joshua Colp : > Ludovic Gasc wrote: > >> 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard > >: >> >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Le 21/05/2015 00:16, Joshua Colp a écrit : >> > If CCSS is needed then the

Re: [asterisk-users] PJSIP CCSS

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 21/05/2015 06:39, Ludovic Gasc a écrit : >> If you really want CCSS support and to be fancy with PJSIP, you can >> easily implement a similar feature with AMI events, I already did tha t a >> long time ago before the integration of CCSS in Asterisk.

Re: [asterisk-users] PJSIP add

Dan Cropp wrote: 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 t

Re: [asterisk-users] PJSIP add

eat day! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, August 25, 2015 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP add

Re: [asterisk-users] PJSIP add

: [asterisk-users] PJSIP add In doing a little research, it seems the Referred-By header could be added after the pjsip_xfer_initiate. This is the approach PJSIP did for some code as far back as PJSIP 1.6. /* * Create REFER request. */ status = pjsip_xfer_initiate(sub, dest

Re: [asterisk-users] PJSIP add

risk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP add Dan Cropp wrote: > 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, every

[asterisk-users] PJSIP Dialout error

Hi Guys I keep getting this "Warning" when I dial out via pjsip and the calls fail But if I do a pjsip reload it works for 1 minute WARNING[6707]: res_pjsip_outbound_authenticator_digest.c:135 digest_create_request_with_auth_from_old: Unable to create request with auth.No auth credentials

[asterisk-users] PJSIP configuration question

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 V

[asterisk-users] PJSIP Bind Issue

Hi, I've got a test server with two IPs (one IP is virtual and moves to a backup server if the first goes down). I'm trying out PJSIP and specified the virtual IP for the Bind address of all transports I'm using. All SIP packets are now sent of the virtual IP. But the RTP UDP data is being sen

[asterisk-users] PJSIP NAT traversal.

I have two servers running pjsip they are both on NAT. The proxy has a static public address. I set the ;external_media_address=203.0.113.1 and ;external_signaling_address=203.0.113.1 to the actual IP address in the transport section on the proxy. The issue I am having is on the server

[asterisk-users] PJSIP Stun/ICE

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind t

[asterisk-users] PJSIP signaling question

Greetings. I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control. I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the existing SIP TLS connection which is already e

[asterisk-users] pjsip segfault problem

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, par

Re: [asterisk-users] PJSIP subscribe

On Mon, Jun 6, 2016 at 11:13 AM, Annus Fictus wrote: > Hello, > > I'm trying to use presence with PJSIP and I have a "issue". > > I created correctly hint priorities like: > > exten => 1000,hint,PJSIP/1000 > exten => 1001,hint,PJSIP/1001 > > Extension 1000 can subscribe extension 1001 y vice-ver

Re: [asterisk-users] PJSIP subscribe

Hello, thank you for the answer... how can I see the correct status? any configuration on asterisk or softphone side? Regards El 07/06/2016 a las 16:36, George Joseph escribió: I can confirm that Bria shows offline but if the client is using the tuple status instead of the person status then

[asterisk-users] PJSIP/Realtime RLS

I see that you can configure RLS in pjsip.conf, but does this work with realtime? The wiki refers to pjsip.conf for configuration, but since many of the other items can be in the the DB, I was wondering if RLS can as well. -- _

[asterisk-users] PJSIP Multipart Body

Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have

[asterisk-users] PJSIP not detected

I installed PJSIP from the project git clone https://github.com/asterisk/pjproject pjproject cd pjproject make uninstall & make distclean ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp make de

[asterisk-users] PJSIP is Ignored

​Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ​checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if

[asterisk-users] PJSIP hints unreliable...

I find that using hints with PJSIP on Asterisk 13 is very unreliable compared to regular SIP. I see many phones as unavailable when they are in fact available. Usually hints will work fine for a while after a phone registers but after a while it will remain at unavailable while it is idle

[asterisk-users] PJSIP missing objects

In version 13.13.0 there is no res_pjsip_keepalive.so res_pjsip_multihomed.so Is this normal? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://com

[asterisk-users] PJSIP logging fails

I am trying to log my SIP registration attempts. PJSIP is in logger mode, and I can see INVITES comingh, my SIP Register does not show, especially the packet I send. The only thing shown is: res_pjsip_outbound_registration.c: No response received from 'snet' on registration attempt to 'sip:7866

Re: [asterisk-users] PJSIP OPTIONS

On Sun, Dec 3, 2017, at 10:34 AM, volga...@networklab.ca wrote: > Hello Everyone, > How to configure PJSIP to reply 200 OK from upstream sip proxy on > keepalive packet ? > > proxy ~> Keepalive OPTIONS ~> asterisk > <~ 200 OK <~ You would need to show the OPTIONS and what is happening now

Re: [asterisk-users] PJSIP OPTIONS

Right now it reply 401 Unauthorized with message in log "No matching endpoint ..." on Content 0 should reply 200 OK I guess <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> OPTIONS sip:10.30.100.27:5080 SIP/2.0 Via: SIP/2.0/UDP 10.30.100.41;branch=z9hG4bKf5eb.1ac7648700

Re: [asterisk-users] PJSIP OPTIONS

On Sun, Dec 3, 2017, at 10:42 AM, volga...@networklab.ca wrote: > Right now it reply 401 Unauthorized with message in log "No matching > endpoint ..." > on Content 0 should reply 200 OK I guess > > <--- Received SIP request (376 bytes) from UDP:10.30.100.41:5060 ---> > OPTIONS sip:10.30.100.27:50

Re: [asterisk-users] PJSIP OPTIONS

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM

Re: [asterisk-users] PJSIP OPTIONS

On Sun, Dec 3, 2017, at 10:55 AM, volga...@networklab.ca wrote: > If understand correctly type=identify is more for sip trunk > configuration ? > > > ;[mytrunk] > ;type=identify > ;endpoint=mytrunk > ;match=198.51.100.1 > ;match=198.51.100.2 > > > In chan_sip it was just reply 200 OK on keep

Re: [asterisk-users] PJSIP OPTIONS

Hello Joshua, What will be example of endpoint configuration that not require authentication from specific ip ? volga629 On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp wrote: On Sun, Dec 3, 2017, at 10:55 AM, volga...@networklab.ca wrote: If understand correctly type=identify is more for sip

Re: [asterisk-users] PJSIP OPTIONS

On Wed, Dec 13, 2017, at 09:38 PM, volga...@networklab.ca wrote: > Hello Joshua, > What will be example of endpoint configuration that not require > authentication from specific ip ? An endpoint doesn't know about an IP address. The identify I previously mentioned is what associates the request t

Re: [asterisk-users] PJSIP OPTIONS

Hello Joshua, Thank you for help that worked. On Thu, 14 Dec, 2017 at 7:06 AM, Joshua Colp wrote: On Wed, Dec 13, 2017, at 09:38 PM, volga...@networklab.ca wrote: Hello Joshua, What will be example of endpoint configuration that not require authentication from specific ip ? An endpoint do

Re: [asterisk-users] PJSIP Originate

On Wed, Mar 14, 2018, at 12:58 PM, Dan Cropp wrote: > I am using AMI Originate to perform a new outbound call. > > The SIP Provider we send the call to wants us to pass the caller id of > the person we are calling for in the Contact header. > > For the AMI Originate, I pass the caller id informa

Re: [asterisk-users] PJSIP Originate

Thanks Joshua -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, March 14, 2018 11:02 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PJSIP Originate On Wed, Mar

[asterisk-users] PJSip CallerID Question

I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from.  I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. For example: Currently Loca

[asterisk-users] pjsip doesn't function

Hi All,I tried to switch from SIP to PJSIP but I can't make any calls. Asterisk 15.4.0Clients: MicroSIP (based on the pjsip SIP stack) With sip.conf all functions OK (SIP instead of PJSIP in extensions.conf) I converted SIP to PJSIP with the script  contrib/scripts/sip_to_pjsip/sip_to_pjsip.py and

[asterisk-users] PJSIP DNS ISSUE

Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but nothing is working. I even installed BIND on the asterisk box ...that didn't even work. Once I pull the plug on the internet, I cant dial anything. John Bittner CTO [xaccell

[asterisk-users] PJSIP IPv6 remote_hosts

Hi, I rey to register an Asterisk 16.2.1 pjsip to an ASTERISK 13.25.0 chan_sip using ipv6 and pjsip_wizard. I only got it work if in remote_hosts I put the ipv6 address and not the hostname like sip.domain.ltd No need to say that an entry is existing for the hostname in DNS. BTW, how

Re: [asterisk-users] PJSIP Qualify

On Sat, Mar 23, 2019, at 9:53 PM, Ian McMaster wrote: > I am currently not using qualify, but it seems like a nice way to know > if the phones are online. I attempted to set it up, but am running into > a 404 on the subscription. > > 1. From the manager, Action: PJSIPNotify (with an endpoint). T

[asterisk-users] PJSIP/SIPAddHeader etc

Hi everyone I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader & SIPRemoveHeader but the apps don’t appear to be installed in v16. Can anyone tell me where they went and how to get them installed please? Thanks Mark. Mark Farmer Senior UC Systems Architect Intercity T

[asterisk-users] pjsip endoint woes

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = [obi202-aor](!) type = aor max_contacts = 2 ; = endpoints [gv-voice](obi202-endpoint) auth = gv-voice aor

Re: [asterisk-users] PJSIP reInvite

Le 15/08/2019 à 13:22, Jöran Vinzens a écrit : Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is ther

Re: [asterisk-users] PJSIP reInvite

Hi, we tried "direct_media=no". this is documented to suppress reInvites but it has no effect. "directmedia" is not known by the config parser and it gives error while reading. direct_media=no is not the same behavior as canreinvite=no, at least as far I can see it. BR Jöran On Thu, Aug 15, 201

Re: [asterisk-users] PJSIP reInvite

Le 15/08/2019 à 14:06, Jöran Vinzens a écrit : Hi, we tried "direct_media=no". this is documented to suppress reInvites but it has no effect. "directmedia" is not known by the config parser and it gives error while reading. Speeking about directmedia was not to point you to a command ;) more

Re: [asterisk-users] PJSIP reInvite

On Thu, Aug 15, 2019, at 8:23 AM, Jöran Vinzens wrote: > Hi All, > > We are using asterisk 16.5 and having an issue with the first re-invite > after the call has been established. > We can see the call gets up and you see in the logs the bridge type has > changed and after that a re-invite is tr

Re: [asterisk-users] PJSIP reInvite

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. He

Re: [asterisk-users] PJSIP reInvite

On Fri, Aug 16, 2019, at 3:06 AM, Jöran Vinzens wrote: > Hi all, > > So the scenario is: > > A -> Asterisk -> B > > after B send back 200 OK Asterisk is answering the call to A. Directly > after the Answer Asterisk generates a ReInvite to A and the only > difference between the 200 OK sdp and

Re: [asterisk-users] PJSIP reInvite

HI All, thanks for your help. We will make our setup work correctly with reInvites. BR Jöran On Fri, Aug 16, 2019 at 11:28 AM Joshua C. Colp wrote: > On Fri, Aug 16, 2019, at 3:06 AM, Jöran Vinzens wrote: > > Hi all, > > > > So the scenario is: > > > > A -> Asterisk -> B > > > > after B send b

Re: [asterisk-users] PJSIP crashes

Did you try to remove the space or replace it (eg underscore) in SIP id likemailto:Dental@8.38.43.67>> ? Le 25/02/2020 à 18:40, Saint Michael a écrit : PJISP cannot handle the From  field when it does not contain a number. Can this be fixed? [Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_tran

Re: [asterisk-users] PJSIP crashes

On 2/25/2020 12:40 PM, Saint Michael wrote: PJISP cannot handle the From  field when it does not contain a number. Sure it can, but: sip:Radefeld Dental@8.38.43.67 Is not a valid SIP URI (it can't contain a space). Is Asterisk actually "crashing" or are you just seeing this error in your log

Re: [asterisk-users] PJSIP crashes

On 2/26/2020 5:06 PM, Saint Michael wrote: PJSIP should log a warting and continue. That's exactly what it is doing unless I am misunderstanding. You didn't answer my question last time - is Asterisk actually "crashing?" It is causing the CPU usage to spike dramatically. If you are able t

Re: [asterisk-users] PJSIP Lockup

On Mon, Mar 2, 2020 at 2:52 PM Nick Olsen wrote: > Hello All, > I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But > recently upgraded to attempt to resolve this issue. Using bundled PJSIP. > The PBX is using mysql realtime for most functions. The Mysql server is on > the same la

Re: [asterisk-users] PJSIP Lockup

Thanks for the info, Joshua. Does PJSIP handle database access the same way Chan_sip did? We had a number of boxes running chan_sip referencing the same mysql server without issue. We're going to attempt to get a backtrace on the next occurance. We're also going to run a local copy of the databas

Re: [asterisk-users] PJSIP Lockup

On Mon, Mar 2, 2020 at 4:24 PM Nick Olsen wrote: > Thanks for the info, Joshua. > > Does PJSIP handle database access the same way Chan_sip did? We had a > number of boxes running chan_sip referencing the same mysql server without > issue. > > We're going to attempt to get a backtrace on the next

Re: [asterisk-users] PJSIP Lockup

We ultimately found this to be a voicemail issue. The voicemail is held in MYSQL as well (via ODBC). And we found when attempting to playback a customers voicemail unavail greeting is when the deadlock would occur (Immediately, every time. Throwing the same "task processors" errors, And making pjsi

Re: [asterisk-users] PJSIP Lockup

- Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP Lockup We ultimately found this to be a voicemail issue. The voicemail is held in MYSQL as well (via ODBC). And we found when attempting to playback a customers voicemail unavail greeting is when the deadlock would occur (Immediately

Re: [asterisk-users] PJSIP Lockup

2020 18:54 > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP Lockup > > We ultimately found this to be a voicemail issue. The voicemail is held in > MYSQL as well (via ODBC). And we found when attempting to playback a > custome

Re: [asterisk-users] PJSIP Lockup

>> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On >> Behalf Of *Nick Olsen >> *Sent:* 01 April 2020 18:54 >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] PJSIP Lockup >> &

[asterisk-users] pjsip subscribecontext support

Hello, I would like to ask about current state of subscribecontext in pjsip. I found out some 6 years old discussion on that without any plans to implement it in the future. I have phones in different contexts. I suspect, when I use its context to subscribe, they will not see phones from the diff

[asterisk-users] pjsip endpoint reachable

hi, we are migrating from chan_sip to pjsip i want logs like this about pjsip endpoints [Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer 'endpoint22' is now Reachable. (15ms / 2000ms) is it possible? thanks Marek -- _ --

Re: [asterisk-users] PJSIP question urgent

CDR wrote: I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)}, so if I use PJSIP, for scalability, how do I read what the signalling IP where the inbound call is coming from and what is the inbound codec? You would think that the new channel would set those variables up, isn't

Re: [asterisk-users] PJSIP Identify Wiky

CDR wrote: The Wiky needs to be updated https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29 This is the example shown: "[6001] endpoint=6001 match=203.0.113.1" It should be: "[6001] type=identify endpoint=6001 mat

Re: [asterisk-users] PJSIP and ARA

Ishfaq Malik wrote: Hi This is a bit of an exploratory question for groundwork before I start playing with asterisk 12. I've spotted the very useful looking file contrib/realtime/mysql/mysql_config.sql in the source. Are the table names starting ps_ all to do with PJSIP? Yes. As well there

Re: [asterisk-users] PJSIP and ARA

On 15 October 2013 14:21, Joshua Colp wrote: > Ishfaq Malik wrote: > >> Hi >> >> This is a bit of an exploratory question for groundwork before I start >> playing with asterisk 12. >> >> I've spotted the very useful looking file >> >> contrib/realtime/mysql/mysql_**config.sql >> >> in the source.

[asterisk-users] PJSIP Include not working

I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled? -- _ -- Bandwidth and Colocation Provided b

[asterisk-users] PJSIP Transfer not working

I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869 pbx_extension_helpe

Re: [asterisk-users] pjsip phoneprov realtime?

On Thu, Nov 13, 2014 at 12:11 PM, John Kiniston wrote: > Howdy, > > Is there a way to use realtime with phoneprov.com and pjsip? > Not yet. I forgot that bit in the initial version of the res_pjsip_phoneprov_provider module. I have a patch ready but it's tangled up in other stuff. I should be

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