similar to: One thread per peer

?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 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 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 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 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?

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.

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 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

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened?

I cannot wait for the regular bug-patch process to play out. I am considering hiring a developer to fix bug 24015, and of course submit the patch for the bug. The issue is simple, the app Transfer does not transfer when using PJSIP.. I called Digium and they said that they do not do this kind of work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

The amount of threads went through the roof ls /proc/15373/task | wc -l 682 in version SVN-branch-12-r427618M it used to be 18 in Asterisk SVN-branch-11-r412226M How can I trace this? There are no calls open, on a disconnected system -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been using the built in Xen Paravirt NIC drivers for paravirt networking and had bad performance until I disabled all offload setting on the bridge and paravirt nic (From Dom0). There are 2 related questions I have, where should I be tweaking the offload settings, the Dom0 Physical NIC, The Bridge or the DomU NIC? Which settings in particular tend to cause trouble? (I just disabled all of

> > ​The community would benefit if a non/licensed version of G729 would be > included with Asterisk​, since the license expired. The current codec > source code posted still requires licensing. > ​I am sure Digium would not prefer to ​ ​acknowledge this, but the phenomenal growth of Asterisk is due to the a​availability of a free G729 codec compiled and distributed free by Arkadi

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

Citrix Project Satori is the result of a collaborative agreement between XenSource and Microsoft, and was carried forward after XenSource was acquired by Citrix Systems. The base Satori components are released by Microsoft as the Linux Integration Components for Hyper-V, and provide support for paravirtualized XenLinux guests running on Hyper-V. The Linux Integration Components can be downloaded

Citrix Project Satori is the result of a collaborative agreement between XenSource and Microsoft, and was carried forward after XenSource was acquired by Citrix Systems. The base Satori components are released by Microsoft as the Linux Integration Components for Hyper-V, and provide support for paravirtualized XenLinux guests running on Hyper-V. The Linux Integration Components can be downloaded

Citrix Project Satori is the result of a collaborative agreement between XenSource and Microsoft, and was carried forward after XenSource was acquired by Citrix Systems. The base Satori components are released by Microsoft as the Linux Integration Components for Hyper-V, and provide support for paravirtualized XenLinux guests running on Hyper-V. The Linux Integration Components can be downloaded