similar to: Need a developer to write me a patch

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?

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

I want to compile Asterisk always for the native architecture of the machine, and I find that it is never available. It says Depends on: native_arch(E) Can use: N/A Conflicts with: N/A Support Level: core This is Fedora 20 gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1) many thanks Philip

?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 "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

I am quite surprised about the degree of surprise in the group. A few days ago, somebody called a school and issued a threat, through my network. The call came from China, but of course it was US caller. The DA wants to know where call came from. The caller ID is "Restricted" and the chinese carrier is playing games. If I had a way to store the media IP, I would be able to pinpoint the

?I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so the Local channel should be able to play anything to the caller, without answering, in parallel

I'm curious about how one would go about contacting/hiring a samba developer to fix a bug, or implement a feature. I've got a particular bug in mind (1493), but it seems like it would be a good thing to know in general. Thanks, Mark Roach

I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command "pjsip reload" was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong?

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

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 SIP call using PJSIP? Philip

My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose

I could not find a way to set a max on the calls allowed through a PJSIP endpoint. In case we decide to add it, the we need another reason for the call to fail in the Dial application, something like "limit reached" Am I missing this capability?

Is there an etiquitte to posting bounties to the centos community for OSS to be packaged in rpm form for centos? -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Principal Consultant 10 West 24th Street #100 - - +1 (443) 269-1555

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hi all, I'll try to explain my suggestion with my poor english. I've got the feeling that icecast is full of potential but not growing as fast as it can or simply not having the support it could have. The greatest lack are the "open doors" for new coders, webdesigners, translators, etc. Icecast is open: the sourcecode is open, the mailing list is open, etc. But a skill is

In channel PJSIP I use this format Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss) what would be the equivalent of this format in old SIP? I tried Dial(SIP/peer/${EXTEN}@ip.add.re.ss) but it does not work. I just cannot embed the IP address in the peer's definition, but I need to use some other configuration features that are unique to each peer.

Is this normal to create one thread per peer in Asterisk 12, chan_sip regular, not pjsip? What happens is I have 659 peers, and I get 682 tasks on ls /proc/15373/task | wc -l If this is normal then of course I can only get a few instances before my box collapses. Is it any different in pjsip? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been seeing errors saying the Asterisk cannot establish an RTP connection, so I did this: lsof -i -n -P | grep asterisk | wc -l 10483 but I have only Asterisk 11 has 1 open calls Asterisk 12 has 21 open calls Asterisk 14 has 19 open calls Asterisk 15 has 22 open calls Asterisk 16 has 15 open calls Asterisk 17 has 15 open calls Asterisk 30 has 71 open calls Total 164 active calls The

We are preparing to build a new Web site and the potential consultant we may be hiring has strongly recommended that we don''t go with our initial planning with asp.net but go with Ruby on Rails. We are not familiar with Ruby but understand their is a strong on-going interest with its development and use. Our concern is that if we need consulting services in the future, are their local

I have the following helper method in application_helper.rb: def format_date(date) day = to_s(date.day) month = to_s(date.month) time = to_s(date.time) date = day + "/" + month + " - " + time return date end I am trying to call this method in a view like this: <%= format_date(bounty.created_on) %> create_on is a timestamp in mysql. I am