similar to: Outgoing trunk is restricted to g.729 but need ulaw

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

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

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

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited)

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

Hello All , In some ocasions i?m getting a problem with my X100P board. I?m trying to trace tre problem , but i didn?t find a possible answer. -> I get those messages when trying to use Zap Channel Sep 29 14:15:46 WARNING[-1094796368]: chan_sip.c:2107 sip_new: Unable to allocate channel structure Sep 29 14:15:46 NOTICE[-1094796368]: chan_sip.c:7283 handle_request: Unable to create/find

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works with Fedora 20. -- Executing [s at DialOut:15] Dial("DAHDI/1-1", "motif/8447/+1212xxxyyyy at voice.google.com,,rTt") in new stack [Mar 1 21:24:06] ERROR[2477][C-00000000]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? [Mar 1 21:24:06]

I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network.

Hi guys I have a site on Asterisk 1.8.11.0 running in Centos 6.5 that has about 150 concurrent callers. I keep getting these types of messages in the CLI: [May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n [May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new:

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

I am trying to run the following model in R > lmer(leaves.eaten~Geocytotype+(1|TEST/ PLANT),data=cyphoplantfeeding,family=poisson) My experimental setup is 41 replicates (TEST) of an experiment in which there are three Geocytotypes of a plant species in each TEST, and two plant pseudoreplicates per Geocytotype in each test (i.e. 3*2=6 plants per test). So my random factors are trying

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -------------- next part -------------- An HTML attachment

Sorry, I sent the following reply with the wrong "from" address and so it did not pass the lists spam filter. So here is the message again: Hi James, Thank you very much for your help! You were right, the codec is not compiled into my asterisk version. I'm using debian, too and a show translation tells me that it is deactivated. Would it be enought to only compile the ilbc codec

Hi: I have a mgcp.conf and a mgcp_additional.conf which records the special information about the extensions. And i found if i use ulaw in the general context in mgcp.conf,then all the registered extensions can make both outbound and inbound calls,the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw ; can be disable and do no effect

I'm running 1.4.22-rc5, and now see the following codec format listed:. What is 0x80004 (ulaw|h) ? 192.168.1.14 101 YjVlYzYwODd 00101/00002 0x4 (ulaw) No Rx: ACK 172.16.1.1 102 7b213e4762c 00102/00000 0x80004 (ulaw|h No Init: INVITE I'm having some voice quality problems and trying to see if it is related to this. Thanks Raja -------------- next part

Simple test extension exten => 14,1,Wait(1) exten => 14,2,SayPhonetic(${CALLERIDNAME}) exten => 14,3,Wait(1) exten => 14,4,SayDigits(${CALLERIDNUM}) exten => 14,5,Hangup Works fine from spa2k extension on lan Works fine calling broadvoice sip did When I call voicepulse sip did I get the calleridname and then silence. CDR logging looks okay but * messages log shows: Sep 19