similar to: Outoing Calls Motif Google Voice Calls Ring After Pick-up

I have an asterisk server at home. I'm looking to replace my internal phones with sip cordless (DECT) phones. I'm now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45). The Siemens has a feature were I can also use a PSTN landline, but I not sure that's a great benefit. Has anybody tried these phones? I

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload => res_jabber.so noload => chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Hi, Are there any thoughts about how "cpu-expensive" motif is? Does it only translate SIP <--> jingle (during call-setup) if so, impact will probably neglectible. or does asterisk remains constantly in between the data-stream? In that case, it might be something to pay serious attention to, when doing multiple call conversions simultaneously... hw

Hello; When I type make menuselect and finding the channels that has the sign XXX before it (this at the driver), how can I know the dependencies that are causing this conflict? Regards Bilal

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

Hello everybody, A client wants to install a FreePBX infrastructure, but have all calls forward to their cell phones rather than buying VoIP phones. They would be doing SIP trunks over a Comcast business line. Probably maximum 6 simultaneous calls. Any gotchas we should warn them about? Thanks! noam Noam Birnbaum El Presidente http://www.desksidemanner.com 415-854-0885 x89 tweet @noamb

Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150117/f148cad7/attachment.html>

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? -- Chris -------------- next part

I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s at xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de

I've now replicated my setup on a host with a single IPv4 address and I am still having trouble with the ICE negotiation. I am trying to call from Jitsi to Asterisk through a Prosody XMPP server. Asterisk successfully registers with the XMPP server and appears to be available in the buddy list in Jitsi. Jitsi is being run with the "-4" command line option to use IPv4 only just in

I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean

Hi, I am very new to R and wanted to know if there is a package that, given very long nucleotide sequences, searches and identifies short (7-10nt) motifs.. I would like to look for enrichment of certain motifs in genomic sequences. I tried using MEME (not an R package, I know), but the online version only allows sequences up to MAX 60000 nucleotides, and that's too short for my needs..

Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hi list , this is a bug? ERROR[361]: res_xmpp.c:3468 xmpp_client_reconnect: No XMPP connection available when trying to connect client regardss -- rickygm http://gnuforever.homelinux.com

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be