Is It Possible To Perform PJSIP Add Header Prior To Calling Queue And Have It Part Of The INVITE Packet?


I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent.

The SIP header I added, I need to have appear in the INVITE sent to the Agent.

It works in chan_sip. I send the call to a macro which does... n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE}) n,Queue(${ARG2})

In PJSIP , this doesn't seem to work. Is there any way to add custom PJSIP headers to be sent as part of the INVITE to the Agent? When I look at the code, it seems as…

Asterisk Users 2 days ago 9 Answers

SIP Trunk - Problem To Connect


Hello! Thnxs for reading! I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset, for instance (and it works!) Connection parameters are:

Authentication Name: Número 1199999999 Authentication password: 12345678 Username: 1199999999 Display name: 1199999999

Domain: Proxy serever address: Proxy server port: 5060 Registrar server: Registrar server port: 5060 Registration refresh time: 1800 Outbound proxy mode: Always Outbound proxy: Outbound proxy port: 5060

Can I add a trunk to my asteriskNOW having only this parameters? I've tested some configurations, but I can find the correct one Could you give me a clue or your point of…

Asterisk Users 3 days ago 0 Answers

Multiple Variable Substitution In Set


Hi All I am trying to do the following: Set(msg=Hello ${world} how ${are} you) I see that ${world} is substituted correctly but not ${are} Using Asterisk 13 I am injecting ${world} and ${are} within an originate action (using Asterisk-Java) I understand one can use max 25 variables in a originate action. I am nowherenear that limit. Any ideas why? Thanksmurthy

Asterisk Users 4 days ago 0 Answers

Pattern Regexten And Dialing To Trunk


Hi list! I'm currently stucked and don't know how to properly resolve next situation:

I'm using asterisk with regcontext and regexten, so far its works: NoOps with prio 1 dynammicaly added or removed as expected.

Example: 3 trunks: tr1 with regexten _1XXX; tr2 with regexten _2XXX and tr3 with regexten _3XXX

'_1XXX' => 1. NoOp(tr1) [SIP] '_1XXX' => 2. Dial(SIP/tr1/${EXTEN}) [pbx_config] '_2XXX' => 1. NoOp(tr2) [SIP] '_2XXX' => 2. Dial(SIP/tr2/${EXTEN}) [pbx_config] '_3XXX' => 1. NoOp(tr3) [SIP] '_3XXX' => 2. Dial(SIP/tr3/${EXTEN}) [pbx_config]

I can't just write _X.,2,Dial(SIP/${EXTEN}) because it is trunks, and i have to write extension with priority 2 for each trunk i…

Asterisk Users 4 days ago 0 Answers

Ringback Issue


My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome.

My situation is that I have many extensions. Here is a sample:

[client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx callerid=D'Arcy <4165555555> mailboxA65555555@VoiceMail context=LocalSets

I can send calls to this extension with this:

exten => 1,Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/4165555555,30) same => n,VoiceMail(4165555555@VoiceMail,u) same => n,Hangup()

Up to this point everything works as expected. I call in and I hear a ringback until the extension is picked up.

Now I add a virtual PBX to the mix.

[pbx-17842] exten =>…

Asterisk Users 5 days ago 3 Answers



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 (verbose 999). In the SIP case, I see it.

Does the function Set(PJSIP_HEADER(add, ..... not transfer over to the call when the Queue…

Asterisk Users 5 days ago 4 Answers