similar to: codec trouble?

Hi folks, In an effort to save bandwidth (our 7905s run over a WAN) we've switched from ulaw to g729a. We purchased 4 licenses from Digium (4 SIP clients, low call volume), and they seem to have been accepted: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 07:53:aa:d3:e2:f2:bd:cc:27:60:9d:5f:da:eb:5d:e9:6e:09:a1:4e == Found license

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/dd10d5ef/attachment.htm -------------- next part -------------- Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this

Hello, Question #1: I have * with g729 installed, and two phones - Cisco 7960 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Hello everyone. I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID provider here in Brazil (GVT - Vox IP service). Here's my scenario: When faxes arrive by a specific DID, they are routed thru this simple macro: [macro-recebefax] exten => s,1,Set(DB(fax/count)=$[${DB(fax/count)} + 1]) exten => s,n,Set(FAXCOUNT=${DB(fax/count)}) exten =>

Ok, My provider is sending a call to me via ULaw but Asterisk isn't picking up on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my sip.conf and that's the only thing I allow, but when my provider sends me the requests, I get an error about No Compatible Codecs: 17 headers, 8 lines Using latest request as basis request Sending to 67.19.245.213 : 5060

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert

Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to

Hi everyone! Who can tell me how I can access my voicemail? When I dial the voicemail on my Cisco 7960 I get access, but when trying to enter my mailbox number it seems that Asterisk doesn't 'get' any of the keys I press. DTMF problem perhaps? Any suggestions on how/where to fix this...? Regards, Evert

Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at