similar to: Asterisk Nat Issue

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

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, the following logs are being generated while i test sip-to-sip windows software phones. Dec 7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call 40dedd1535853f17250b4d0854e35c17@200.75.243.237 for seqno 102 (Critical Request) == No one is available to answer at this time Dec 7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt: Maximum

Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Is there a way to increase the number of retries or the time to help with this? WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103 (Request) WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for seqno 103

The following message occurs every two minutes in my log file when in diagnostic mode. Aug 22 13:58:00 WARNING[1133735216]: File chan_sip.c, Line 421 (retrans_pkt): Maximum retries exceeded on call 46afc0866a7987591248e29d299e47ee@10.1.1.63 for seqno 102 (Request) Aug 22 14:00:01 WARNING[1133735216]: File chan_sip.c, Line 421 (retrans_pkt): Maximum retries exceeded on call

Hello list! I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on

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

Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone

Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: >Pamela, >Did you resolve the problems you described? >I didn't see a reply on the list but I may have missed it. > >-Kevin > >-----Original Message----- >From: Pamela Weis [mailto:peawy@gmx.at] >Sent: Thursday, August 05, 2004

Hi, After a few attempts, I've managed to grab the files from CVS and build it on a rh redora box I've setup especially for Asterisk. Firstly, we're new to the asterisk scene, so please excuse any "lame" questions which may follow.. We're a new voiptalk.org customer. We have purchased the voip phones (budgetone 102's) and set aside a little box to run Asterisk on.

Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

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

Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and