similar to: chan_sip loading delay in Asterisk 1.2.10

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Hello, So, I found this line under macro-dialout-trunk, in extensions_additional.conf (FreePBX, so it controls the conf files mostly); exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) If I grep for OUT_3_SUFFIX in all files in /etc/asterisk I get nothing.. Here's a paste of a few things out of the two files that I

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

This is a small 12 line system, internal extensions 150 - 180. I didn't have a phone on 151. Here's the sip.conf stanza: ;;[151] ;;type=friend ;;context=longdistance ;;callerid="Conf Room" <151> ;;secret=0000 ;;host=dynamic ;;qualify=yes ;;dtmfmode=rfc2833 ;;allow=all ;;defaultuser=151 ;;nat=yes ;;canreinvite=no There's no DISA. And then somehow (how???) ip address

I currently have the following outbound-local config in my setup.... I can call SOME of the numbers (like 337xxxx, and 998xxxx, and 323xxxx).. but when I try to dial say like 601xxxx I get a 404.. any thoughts, I can't see any difference in the config. Also, I seem to be able to dial any number that starts with a 9.. such as 977, 990, 903.. [outbound-local] ;exten =>

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

I tried adding wwww to my dial string... It appears to not made any differnece. I see by watching show channels that the w's are there in the dial. I have this box connected to bell south down in georgia. I have an identical box connected in indiana. The box in indiana works fine when dialing out. When in GA dialout is getting "you must first dial a 1 to place this call". show

I have a call from a call file: Action: Originate Async: yes Channel: SIP/2012 Codecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 20000 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

Yes indeed. Do you have the dialplan, eg from /etc/asterisk/extensions.conf? Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 172.22.4.12. On 18 August 2015 at 16:21, Brendan Ord <bord at staff.onthenet.com.au> wrote: > Starting to make sense when I saw this line: > > > > [2015-08-18 15:01:33] DEBUG[19366][C-00001cfc]: pbx.c:3785 >

El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. > > Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at