similar to: PJSIP vs SIP channeltype

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones? For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP. On each of those servers, there are a mix of SIP and SCCP phones attached. Currently, I'm using res_xmpp to distribute mwi from the voicemail server to the endpoint servers. Would this type of setup work

I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a

[this went to me instead of the list; MM, your list maintainer] I am digging in the wrong hole I guess. Where can I find R /S routines for spatial statistics? Also has anyone made an R link to a GIS package? Anyone out there who works in this area? Thanks Richard E. Hoskins WA State Department of Health 1102 Quince Street Olympia, WA 98504-7812 richard.hoskins at doh.wa.gov tel: (360) 236 -

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

Good Morning, After further investigation, I found that the res_pjsip_publish_asterisk module does not use the realtime sorcery wizard, but instead only reads from the configuration files. I've been able to patch the module, using the logic from the other modules to learn how to make the sorery configuration read from the other sorcery wizards and it's now working for the

Joshua, Regarding Outbound Registrations in realtime and a reload. Does it require a "pjsip reload" or full asterisk restart? Antonio G?mez Soto wrote: > Hi, > > The asterisk wiki page says: > > "Sorcery.conf allows you to try to configure other PJSIP objects such > as transport using realtime and it currently won't stop you from doing so. > However,

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions

Hi, I have a data frame with two columns of data, one an indexing column and the other a data column. My issue is, this data frame is incomplete and there are missing lines. I want to know how I can find and add data into these missing lines. See example below ## Example data data <- data.frame(index=c(1:4,6:10), data=c (1.5,4.3,5.6,6.7,7.1,12.5,14.5,16.8,3.4)) index data 1 1

Hello, I have a problem width some excel files in my Samba File server because when I try to open from a WinMe client, the client machine always craches width a error in the vredir.vxd and ifsmgr.vxd. I think is a problem in oplocks or something like that. In the mdb files, the MS Access tolds me that can't open the file exlusively! Can anybody help me? What configuration should I have in

Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the

Hi there, I almot exlusively use samba for printing in a semi large scale (>2.000 windows hosts & >500 printers). I have been running 3.0.20 for a long time and was quite happy with it. Unfortunately my windows fellows have now updated to AD 2008 R2 and I experienced some immediate problems with encryptions types so I updated to 3.4.5 which is not really working for me. I see a

Hi there, I almot exlusively use samba for printing in a semi large scale (>2.000 windows hosts & >500 printers). I have been running 3.0.20 for a long time and was quite happy with it. Unfortunately my windows fellows have now updated to AD 2008 R2 and I experienced some immediate problems with encryptions types so I updated to 3.4.5 which is not really working for me. I see a

I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to

This is not specifically related to * but * is the software I'm using so here goes... Does anyone have the correct file set for a 7960?? I've been trying to get the release 6 SIP load on one I have without any luck. The phone keeps getting the same 2 files from the tftp server and starting over. If you have the files - other than the POS30600.bin which I know is licensed - could you

Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet

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