similar to: Maximum retries exceeded on transmission

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hello All, We are running Asterisk on a linux server as a SIP proxy with Cisco ATA 186's at the subscriber end. For long distance we have iax2 connectivity with a ip carrier. For local calls we are routing out through a commercial VEGA voicestream pots unit to an adtran channel bank and then from there to our class 5 soft switch. The sip to sip calls and the long distance calls work great.

I have an Asterisk 1.4 box which is sometimes getting the message below. Here is the weird part, the CNG is coming from ANOTHER ASTERISK SERVER. 209.220.119.19 is an Asterisk 11 box. [Oct 8 11:59:27] NOTICE[20798]: rtp.c:849 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 209.220.119.19

Hi guys, I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk. Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not. Then I saw that message appear on the Asterisk CLI, during

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!

Hi, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

Hi Everyone, I was hoping someone might know why I am experiencing a problem with Asterisk logging the event: [May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission 03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com for seqno 669371069 (Critical Response) This is happening after: - call is setup, 2 way audio - call can function correctly for up to 5

hello to all, i am geting this NOTICE while i am running asterisk. agents are able to here the customer voice but the customer is unable to here agent voice plz somebody help me #rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 -- M. VIDYASAGAR -------------- next part -------------- An HTML

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

When making or receiving a SIP call via my service provider, I get the following message logged by Asterisk: Dec 11 15:13:37 NOTICE[7392]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Since the "client" is at my service provider (who uses CISCO kit, I believe), I don't have the

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.