similar to: stuck on retransmit

Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and only in voicemail) after about 20 seconds with the error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt).". I

? HI ? I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) ? Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg ? Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858

Hello, Please forgive me if this has been addressed previously. I have been searching the archives and have not come across what I thought was a solution. My * server is behind a DSL router using a NAT IP address of 10.0.0.9. A colleague running XP and X-Lite can register with * from his home, specifying my public IP as the SIP proxy in X-Lite (however this is only true if I have the NAT flag

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

Hi, I've got Asterisk configured and working (sort of) with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). This * box is on a 'live', non-nat IP address. I also have a couple of budgetone phones, one behind NAT and one not. When I place an outgoing call, I get the following messages: -- Executing Dial("SIP/filbert-9876", "CAPI/288:333") in

Hello all, Over the weekend, I setup and linked an Asterisk box at another site to the Asterisk box here. The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100 phones. The phones at the other end are Grandstream BT-100 SIP phones. The Cisco phones run SIP 7.1 (upgraded last Friday from 6.1), the Grandstream phones run 1.0.4.68. Both Asterisk boxes are running stable CVS

Skipped content of type multipart/alternative-------------- next part -------------- Reliably Transmitting (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact:

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hey Peoples, I just got my paws on a KE1020A Phone and all it is doing when I plug it in is: 1201 Wait Login... Sip.conf [1201] type=friend username=1201 secret=<password> host=216.254.10.183 mailbox=1201 context=intern canreinvite=yes dtmfmode=rfc2833 nat=1 register => 1201:<password>@216.254.10.183/1201 One side note, The KE1020A does not have NAT capabilities, but I am

Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged

when I make a call to the pstn it shows me this error: aximum retries exceeded on transmission 9d4a24f8-b673756b at 192.168.10.19 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Apr 11 20:35:34] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging up call 9d4a24f8-b673756b at 192.168.10.19 - no reply to our critical packet (see doc/sip-retransmit.txt). bug? voicemail same

se?ores alguien le ha presentado este problema al acceder al voicemail o al hacer una llamada a la pstn 1940> Playing 'vm-received' (language 'es') -- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es') -- <SIP/111-08d91940> Playing 'digits/at' (language 'es') -- <SIP/111-08d91940> Playing

What is the problem with SIP retransmits? ----------------------------------------- Sometimes you get messages in the console like these: - "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet." - "retrans_pkt: Cancelling retransmit of OPTIONs" The SIP protocol is based on requests and replies. Both sides send requests and wait for replies.

Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]:

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

Hi All, I'm currently having trouble with dropped phone calls. The following error message is always in the log. This is a Grandstream GXP-2000 Firmware 1.1.6.16 . The Asterisk box is currently 1.4.22-rc5. The problem has been occurring on other versions also. [Sep 19 15:48:02] WARNING[13657]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission 8acaea6dc4c6e9b5 at