similar to: SIP HEADER FROM: without CALLERID(name):: PART DEUX

Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of

I have added a 4th NIC to my setup, and want to set up wireless. I have stared at the configuration Tom has for the last week, and my eyes are crossing. eth0 "net" goes to my internet connected firewall with a 192.168 address eth1 "loc" goes to my switch connected to local switch also 192.168.x eth2 "work" goes to my office with a 172. address eth3 Trying to

I was wondering how to get access to the headers of the INVITE on incomming SIP calls in the Dialplan. My scenario is that i use "register" in sip.conf to register a UA on which to accept incomming calls. In sip.conf, calls to that UA is redirected to a specific extension in extensions.conf (btw: can that be a dynamic value, for example ${CALLERID} or

Hello, I've added a ZAPHFC card to my CAPI based system. Calls coming in via ZAPHFC do not forward the caller id to the SIP phones. Calls coming in via CAPI do forward the caller id to the SIP phones. Any and all help is greatly appreciated. The (hopefully relevant) conf file excerpts are: extensions.conf =============== exten => 807440,1,Answer exten => 807440,2,Noop exten =>

Hi! I trying to configure * in a way, that it uses a different CLIP (Caller-Id in ISDN) per SIP user, when relaying the call from SIP to the ISDN. So far always the main (1st) number of the number-block is sent to the ISDN. I have a E100P from Digium and use the zapata stuff (chan_zap). All SIP calls are coming through an SER. One idea I had in mind is to assign userid's in SIP, that match

Hi all ! First, my apologies if this subject has already been discussed but the archives lack a "search" feature, so I've browsed a bit but not a lot before 2001. I browsed the winehq and I found some people have been able to use wine on Deux Ex, and Baldur 1 and 2. I installed the games with windows (they worked fine) and then tried running them with wine under linux (debian

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hello all, My pbx server is connected to a sip gateway, when I call an originate command from the asterisk console, to establish a sip connection, the gateway doesn't accept URL with white spaces, for example: * Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport * * From: "PBX SERVER" <sip:PBX SERVER at 10.10.1.10>;tag=as2512881b * * To:

Hello friends. I have a softswitch in which I cannot create a list of blocked source numbers; So, I have thought to use Asterisk and return a 302 message when the number can make the call, my dialplan is as follows: [from-external]   exten => _AX.,1,Verbose(=======> ${CALLERID(num)} to ${EXTEN})    same =>      n,Set(MYDESTINY=${REPLACE(${EXTEN},A,)})    same =>     

Hi, I have two asterisk servers with the same version of 1.4.29.1. The first server named it as MYE1. MYE1 is an incoming server that can accept incoming calls from PSTN(ZAP E1). The second server is a pbx functions server and named it as MYPBX(SIP). The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

I have been unable to get callerid name passed from Cisco Callmanager over a SIP trunk to Asterisk. Only the number is displayed. Has anyone been successful getting callerid name?

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Is there a way I can forward to my phones the domain of the CALLERID in the CALLERID(number) field of INVITE messages, when some call arrives to my Asterisk? What happens in my architecture is this: INVITE john at our_mydomain INVITE john at phone's_IP ------------------------------------------------------->

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

Curious if anyone else has run into this. I am testing this with a Sipura-2000 with firmware rev 1.0.33. On FC1 with Asterisk 1.0 Stable branch 4/1/2004 CVS (no April fools jokes please :) ) The Sipura has the ability so when you dial *67 it turns ON CID block and *68 turns it back off. (This is for outbound calls) When I *67 (activate CID Block) dial, and look at the SIP INVITE I see that it

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

I am having an issue with the CallerID Number not being passed to my phone in the SIP packet. The CallerID Name is passed just fine and displayed on the phone with no issue. I have done a NoOp() in my extension.conf and successfully seen both the CallerID name and number correctly. So that leads me to believe that Asterisk is handeling it correctly. However, when I do a packet capture of the

Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication "407 Proxy Authentication Required". It seems that the

Is it possible to modify the caller id on the phone during a call (session) ? If not does anybody know with which SIP request this could be handled ? I'm know investigating RFC3311 which seems to offer an answer but if somebody already has an answer ... Michael

OK, I've been banging my head on why my procmail setup for virtual users is no longer working (difficult to test, since enabling it breaks live user's mail). There are only a few virtual users who have any sort of filters in place anyway (the heavy procmail users are local, not virtual), and they are fairly simple, so I think I can recreate them with sieve. I think I have everything I