# Uncategorized **URL:** https://community.asterisk.org/c/uncategorized/1.md?page=2 [Latest](https://community.asterisk.org/latest.md) · [Categories](https://community.asterisk.org/categories.md) **Page:** 3 --- ## [Call establishment through fake ip address](https://community.asterisk.org/t/call-establishment-through-fake-ip-address/67598)
**Author:** [@LemosSIP](https://community.asterisk.org/u/LemosSIP)\ **Replies:** 1\ **Last updated:** [August 6, 2016, 4:06am UTC](https://community.asterisk.org/t/call-establishment-through-fake-ip-address/67598 "2016-08-06T04:06:53Z")
Hi everyone, is possible establish a voip call from SIP INVITE message carried by a udp packet with fake ip address? Is more likely that the replies will be sent to the fake address, so the call cannot establish communi… --- ## [Asterisk as SIP registrar for CMR Cloud](https://community.asterisk.org/t/asterisk-as-sip-registrar-for-cmr-cloud/67603)
**Author:** [@Ciag](https://community.asterisk.org/u/Ciag)\ **Replies:** 1\ **Last updated:** [August 5, 2016, 5:16pm UTC](https://community.asterisk.org/t/asterisk-as-sip-registrar-for-cmr-cloud/67603 "2016-08-05T17:16:05Z")
Hi Folks, I'm a complete noob here so I'm sure my question will be easily answered. I'm testing Cisco webex CMR cloud at the moment, one of the prerequisites for video conferencing endpoints to connect to the CMR cloud… --- ## [Add debug output to app module code](https://community.asterisk.org/t/add-debug-output-to-app-module-code/67610)
**Author:** [@redwolf890](https://community.asterisk.org/u/redwolf890)\ **Replies:** 3\ **Last updated:** [August 5, 2016, 12:20pm UTC](https://community.asterisk.org/t/add-debug-output-to-app-module-code/67610 "2016-08-05T12:20:13Z")
I'm playing with modifying Asterisk 11.19.0's app\_voicemail.c code. Specifically I want to modify one of the message features (forward voicemail option 8). To start with I am going to just use some of the variable valu… --- ## [Out Call failed with ISDN](https://community.asterisk.org/t/out-call-failed-with-isdn/67617)
**Author:** [@jaspreet\_Singh](https://community.asterisk.org/u/jaspreet_Singh)\ **Replies:** 0\ **Last updated:** [August 5, 2016, 11:43am UTC](https://community.asterisk.org/t/out-call-failed-with-isdn/67617 "2016-08-05T11:43:52Z")
Hi Team, I am trying to make out call with .call file but getting error.Please help. Error:- pbx\_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO\_ANSWER, maybe Circuit busy or down?) … --- ## [Log File Time Stamps](https://community.asterisk.org/t/log-file-time-stamps/67611)
**Author:** [@TommyFrass1](https://community.asterisk.org/u/TommyFrass1)\ **Replies:** 1\ **Last updated:** [August 5, 2016, 6:51am UTC](https://community.asterisk.org/t/log-file-time-stamps/67611 "2016-08-05T06:51:16Z")
Can anyone help me time stamp my log files? For whatever reason it does not seem to be working --- ## [Musiconhold With Ringing Sound](https://community.asterisk.org/t/musiconhold-with-ringing-sound/67516)
**Author:** [@barslan2](https://community.asterisk.org/u/barslan2)\ **Replies:** 4\ **Last updated:** [August 5, 2016, 3:43am UTC](https://community.asterisk.org/t/musiconhold-with-ringing-sound/67516 "2016-08-05T03:43:11Z")
Hi , When I try to call someone I want to hear musiconhold with ringing sound. My dial plan on below This dial plan is working but I can't hear ringing sound. Do you have any idea about it? exten =\>978,1,Set(CHANN… --- ## [Flood of UDP packets Len=172](https://community.asterisk.org/t/flood-of-udp-packets-len-172/67609)
**Author:** [@Syco](https://community.asterisk.org/u/Syco)\ **Replies:** 1\ **Last updated:** [August 4, 2016, 3:50pm UTC](https://community.asterisk.org/t/flood-of-udp-packets-len-172/67609 "2016-08-04T15:50:40Z")
Hello I have a question that might be very stupid, but I can't find an answer anywhere. A client has been complaining that we flood him with udp packets. I've got one packet and its reply and pasted it to http://pastebin… --- ## [Tap incoming audio to softphone/speaker](https://community.asterisk.org/t/tap-incoming-audio-to-softphone-speaker/67608)
**Author:** [@satish.setty](https://community.asterisk.org/u/satish.setty)\ **Replies:** 0\ **Last updated:** [August 4, 2016, 12:16pm UTC](https://community.asterisk.org/t/tap-incoming-audio-to-softphone-speaker/67608 "2016-08-04T12:16:30Z")
Hello, We are trying to convert speech (in English language) to text in real time but facing issues. We have a "softphone - ip based" installed on pc. Calls received by softphone needs to be converted to text. Guess au… --- ## [AMI 2.6.0 displays PeerStatus events twice in a row](https://community.asterisk.org/t/ami-2-6-0-displays-peerstatus-events-twice-in-a-row/67580)
**Author:** [@snishizaki](https://community.asterisk.org/u/snishizaki)\ **Replies:** 2\ **Last updated:** [August 3, 2016, 4:47pm UTC](https://community.asterisk.org/t/ami-2-6-0-displays-peerstatus-events-twice-in-a-row/67580 "2016-08-03T16:47:06Z")
I am running Asterisk 13.1.0 with AMI 2.6.0 Every time that Asterisk throws a PeerStatus event (not when called with a peerStatus action), the event is generated twice in a row with sequential sequence numbers and times… --- ## [Asterisk and WebRTC](https://community.asterisk.org/t/asterisk-and-webrtc/67596)
**Author:** [@rayj00](https://community.asterisk.org/u/rayj00)\ **Replies:** 0\ **Last updated:** [August 3, 2016, 4:16pm UTC](https://community.asterisk.org/t/asterisk-and-webrtc/67596 "2016-08-03T16:16:40Z")
So, I am wondering what the state of WebRTC and asterisk is? I would like to use asterisk strictly for WebRTC video and audio calls. Does the current Asterisk support WebRTC fully? Is there an html based softphone for … --- ## [Writing a CTI application](https://community.asterisk.org/t/writing-a-cti-application/67569)
**Author:** [@astbox](https://community.asterisk.org/u/astbox)\ **Replies:** 1\ **Last updated:** [August 3, 2016, 4:02pm UTC](https://community.asterisk.org/t/writing-a-cti-application/67569 "2016-08-03T16:02:48Z")
Planning to write a CTI application for Windows, what is the best interface to choose, the old and trusted AMI or should I begin with the "new" rest api that asterisk supports? Can asterisk through ARI feed my client wit… --- ## [WFM and Call Recording integration](https://community.asterisk.org/t/wfm-and-call-recording-integration/67588)
**Author:** [@Martok](https://community.asterisk.org/u/Martok)\ **Replies:** 3\ **Last updated:** [August 3, 2016, 11:54am UTC](https://community.asterisk.org/t/wfm-and-call-recording-integration/67588 "2016-08-03T11:54:26Z")
Hi all, can anyone help me with the best to start looking for information on integrating 3rd party systems to an Asterisk NOW platform? In particular I am interested in WFM and call recording. Any pointers would be very … --- ## [PJSIP and realtime](https://community.asterisk.org/t/pjsip-and-realtime/67593)
**Author:** [@enzodd](https://community.asterisk.org/u/enzodd)\ **Replies:** 11\ **Last updated:** [August 3, 2016, 11:33am UTC](https://community.asterisk.org/t/pjsip-and-realtime/67593 "2016-08-03T11:33:10Z")
Hi to all, i have a problem ( :frowning: ) I've configured realtime, odbc connection and all works properly. But when i do "pjsip show endpoints", the output message is "No objects found.". Endpoints are present into d… --- ## [Allow Header missing UPDATE](https://community.asterisk.org/t/allow-header-missing-update/66862)
**Author:** [@Novais](https://community.asterisk.org/u/Novais)\ **Replies:** 18\ **Last updated:** [August 3, 2016, 7:24am UTC](https://community.asterisk.org/t/allow-header-missing-update/66862 "2016-08-03T07:24:00Z")
We are having an issue with calls dropping from one of our Trunks. We have run a SIP debug and find that there is a timeout on a non-critical invite. During one of the INVITE transactions we are not seeing a response f… --- ## [Asterisk for Voice Call Conference](https://community.asterisk.org/t/asterisk-for-voice-call-conference/67590)
**Author:** [@itsupport.kipl](https://community.asterisk.org/u/itsupport.kipl)\ **Replies:** 2\ **Last updated:** [August 3, 2016, 4:54am UTC](https://community.asterisk.org/t/asterisk-for-voice-call-conference/67590 "2016-08-03T04:54:10Z")
Hi All, We are new to Call Conferencing, hence in this regards we need help for Conference call for at-least 25 peoples. So, in this regards can anybody let me know exactly which Asterisk product we required & what wil… --- ## [Asterisk "meet me" functionality](https://community.asterisk.org/t/asterisk-meet-me-functionality/67592)
**Author:** [@carndt](https://community.asterisk.org/u/carndt)\ **Replies:** 1\ **Last updated:** [August 3, 2016, 4:45am UTC](https://community.asterisk.org/t/asterisk-meet-me-functionality/67592 "2016-08-03T04:45:18Z")
I was recently asked if asterisk could provide this... Quoting asker: I'd like it to notify us (text? Email?) after 2 mins, 5 mins, and then every 10 mins hold time. Added bonus if I can call in to answer the call ("m… --- ## [Asterisk 11: Can't record calls in queues](https://community.asterisk.org/t/asterisk-11-cant-record-calls-in-queues/66860)
**Author:** [@gromm](https://community.asterisk.org/u/gromm)\ **Replies:** 16\ **Last updated:** [August 2, 2016, 9:38pm UTC](https://community.asterisk.org/t/asterisk-11-cant-record-calls-in-queues/66860 "2016-08-02T21:38:21Z")
We're having some trouble with Asterisk v11.13.1 According to this document http://www.asteriskdocs.org/en/3rd\_Edition/asterisk-book-html-chunk/ACD\_id288901.html#options\_defined\_queues about the only thing you need to … --- ## [Asterisk ODBC works, but Realtime SIP does not](https://community.asterisk.org/t/asterisk-odbc-works-but-realtime-sip-does-not/67549)
**Author:** [@gromm](https://community.asterisk.org/u/gromm)\ **Replies:** 7\ **Last updated:** [August 2, 2016, 7:27pm UTC](https://community.asterisk.org/t/asterisk-odbc-works-but-realtime-sip-does-not/67549 "2016-08-02T19:27:05Z")
Hi everyone. I'm having trouble getting SIP to recognise the Realtime users we have in our MySQL database. ODBC works fine within Asterisk 13.1-cert7, so I'm not going to go over that configuration: template25\*CLI\> odb… --- ## [Asterisk test audio call](https://community.asterisk.org/t/asterisk-test-audio-call/67578)
**Author:** [@tanio.artino](https://community.asterisk.org/u/tanio.artino)\ **Replies:** 2\ **Last updated:** [August 1, 2016, 1:43pm UTC](https://community.asterisk.org/t/asterisk-test-audio-call/67578 "2016-08-01T13:43:37Z")
Is it a way to do the following in asterisk Asterisk answers a call The user says a message Asterisk repeats the message Like what Skype does for testing the audio devices Thanks --- ## [Telecom Engineer Job in Seattle](https://community.asterisk.org/t/telecom-engineer-job-in-seattle/67575)
**Author:** [@procvar](https://community.asterisk.org/u/procvar)\ **Replies:** 0\ **Last updated:** [August 1, 2016, 5:50am UTC](https://community.asterisk.org/t/telecom-engineer-job-in-seattle/67575 "2016-08-01T05:50:24Z")
Asterisk experience is very desirable! DESCRIPTION We are seeking a VoIP telecom engineer to help us design, evolve, and support the telephony network that powers the world's #1 customer service team. Our network han… --- ## [Grandstream GXV3175v2](https://community.asterisk.org/t/grandstream-gxv3175v2/67567)
**Author:** [@sbsinfo](https://community.asterisk.org/u/sbsinfo)\ **Replies:** 1\ **Last updated:** [July 31, 2016, 2:33pm UTC](https://community.asterisk.org/t/grandstream-gxv3175v2/67567 "2016-07-31T14:33:42Z")
hello i have some problem to connect a grandtream GXV 3175v2 over openvpn to my freepbx server Current Asterisk Version: 13.9.1 PBX Firmware: 10.13.66-13 openvpn is connect but the phone doesn't register can someone hel… --- ## [Configuring or changing current users](https://community.asterisk.org/t/configuring-or-changing-current-users/67536)
**Author:** [@TS1](https://community.asterisk.org/u/TS1)\ **Replies:** 1\ **Last updated:** [July 27, 2016, 8:06pm UTC](https://community.asterisk.org/t/configuring-or-changing-current-users/67536 "2016-07-27T20:06:33Z")
Hi, I inherited this system with no training and a manual that is missing most commands.. I am trying to find out how to make changes to current users phones. I can get to a command line but when I try to use any of the… --- ## [asteriskNow Distro with CTI or screenpop](https://community.asterisk.org/t/asterisknow-distro-with-cti-or-screenpop/67446)
**Author:** [@chasemixon](https://community.asterisk.org/u/chasemixon)\ **Replies:** 4\ **Last updated:** [July 27, 2016, 10:46am UTC](https://community.asterisk.org/t/asterisknow-distro-with-cti-or-screenpop/67446 "2016-07-27T10:46:52Z")
Has anyone implemented CTI with an asteriskNow Distro? I found this http://nerdvittles.com/?p=234 and modified my configuration according to the instructions knowing that mine would be a little different, to modify my ht… --- ## [PJSIP, sorcery, realtime & res\_pjsip\_phoneprov\_provider: phoneprov objects in realtime db?](https://community.asterisk.org/t/pjsip-sorcery-realtime-res-pjsip-phoneprov-provider-phoneprov-objects-in-realtime-db/67529)
**Author:** [@earlybird](https://community.asterisk.org/u/earlybird)\ **Replies:** 2\ **Last updated:** [July 27, 2016, 9:52am UTC](https://community.asterisk.org/t/pjsip-sorcery-realtime-res-pjsip-phoneprov-provider-phoneprov-objects-in-realtime-db/67529 "2016-07-27T09:52:07Z")
Currently, the documentation for PJSIP realtime in the wiki does not document storing objects of type=phoneprov in the realtime database, nor does sorcery.conf.sample document this object type, and nor do the alembic mig… --- ## [Asterisk Trunk Transfer Problem?](https://community.asterisk.org/t/asterisk-trunk-transfer-problem/67518)
**Author:** [@the\_passerby](https://community.asterisk.org/u/the_passerby)\ **Replies:** 5\ **Last updated:** [July 27, 2016, 5:12am UTC](https://community.asterisk.org/t/asterisk-trunk-transfer-problem/67518 "2016-07-27T05:12:26Z")
\[image\] Sorry for miss writing on image. 500 calls 502, speaks with 502 without any problem then 500 pushes transfer button and calls 503, after that 500 hears 502's voice, 503 hears 500's voice. I tried this scenar… --- ## [SDP packet sendonly and Connection Information 0.0.0.0?](https://community.asterisk.org/t/sdp-packet-sendonly-and-connection-information-0-0-0-0/67514)
**Author:** [@the\_passerby](https://community.asterisk.org/u/the_passerby)\ **Replies:** 9\ **Last updated:** [July 26, 2016, 12:29pm UTC](https://community.asterisk.org/t/sdp-packet-sendonly-and-connection-information-0-0-0-0/67514 "2016-07-26T12:29:09Z")
In transfer scenario transferror send SIP/SDP packet. In SDP side, transferror puts "sendonly" and "Connection Information IP" as 0.0.0.0 and sends this packet to transferee. Why transferror sends "sendonly" to transfer… --- ## [Lots of short sound files kills asterisk](https://community.asterisk.org/t/lots-of-short-sound-files-kills-asterisk/67510)
**Author:** [@thecodeguy](https://community.asterisk.org/u/thecodeguy)\ **Replies:** 0\ **Last updated:** [July 25, 2016, 10:28pm UTC](https://community.asterisk.org/t/lots-of-short-sound-files-kills-asterisk/67510 "2016-07-25T22:28:16Z")
Is there a reason that lots of short sound files played consecutively would cause asterisk to crawl and ultimately stop answering calls? If I set a 5 second file, followed by a 10 second silence file and repeat, the cpu… --- ## [Add function in confbridge which on (9) change default\_user to admin\_user in a conference](https://community.asterisk.org/t/add-function-in-confbridge-which-on-9-change-default-user-to-admin-user-in-a-conference/67474)
**Author:** [@ahmed\_siddique](https://community.asterisk.org/u/ahmed_siddique)\ **Replies:** 1\ **Last updated:** [July 25, 2016, 12:49pm UTC](https://community.asterisk.org/t/add-function-in-confbridge-which-on-9-change-default-user-to-admin-user-in-a-conference/67474 "2016-07-25T12:49:46Z")
hi all, First of all im really sorry im new here so i dont know much about asking for Help but here it is , I need help with my dial plan i want to make a default user a admin on a digit press for example when i press… --- ## [Asterisk return value to external file](https://community.asterisk.org/t/asterisk-return-value-to-external-file/67493)
**Author:** [@dr.x](https://community.asterisk.org/u/dr.x)\ **Replies:** 4\ **Last updated:** [July 24, 2016, 11:05pm UTC](https://community.asterisk.org/t/asterisk-return-value-to-external-file/67493 "2016-07-24T23:05:04Z")
Hi say that there is a call and working into a context . i want to add some rules inside the dial plan to extract some asterisk variables like CID , DID to external file . say there was a call . i want to do like ec… --- ## [Queues for call files?](https://community.asterisk.org/t/queues-for-call-files/67485)
**Author:** [@softfarr](https://community.asterisk.org/u/softfarr)\ **Replies:** 2\ **Last updated:** [July 23, 2016, 8:19am UTC](https://community.asterisk.org/t/queues-for-call-files/67485 "2016-07-23T08:19:55Z")
Hi. Because I'm newbe I don´t know how call files work and I need some feedback. I'm working in an automatic dialer on demand via shell scripts on asterisk 11 and CentOS and one SIP channel based in PSTN line. It´s in … [Previous page](https://community.asterisk.org/c/uncategorized/1.md?page=1) [Next page](https://community.asterisk.org/c/uncategorized/1.md?page=3)