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 17 16:37:47 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 102 (Request) Mar 17 16:43:43 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 0162f8c72fdabef42b09d61d49db3092@66.92.222.38 for seqno 102 (Request) Setup is OBSD 3.4, asterisk 0.7.2, X-Lite softphone, and Cisco ATA186. PF is disabled entirely, and all nodes are behind the same NAT, on the same LAN. I see all kinds of posts in the archives with this problem, but no clear solution. Suggestions? Regards, Ed Hintz ed@hintz.org
check NAT setting try taking it out of sip.conf, that worked for me Barry
Eric C. Snowdeal III
2004-Jun-17 18:09 UTC
[Asterisk-Users] Maximum retries exceeded on call
i'm new to asterisk and am having trouble placing outbound calls. i know this topic has been discussed ad nauseum in the past [1] , but i can't seem to find a workaround and i'm wondering if my newbie-ness is getting the best of me. after registering the phones correctly and receiving a "200 o.k." message i can connect to other registered softphones and pstn endpoints [ via an voicepulse account ], but after making the initial connection, i can't hear any sound and i get disconnected after getting the following error message: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 0F2F6379-C0BD-11D8-BF01-000D93C22AF4@192.168.1.100 for seqno 41270 i've compiled the stock asterisk tarball on a redhat 7.3 box with a public ip address. the clients are xten lite softphone's running on ibooks with os 10.3.4. the clients are natted behind a linksys wrt54g wireless router running the sveasoft [2] firmware. i'm perplexed, because i can get things to work fine if i use ser/rtpproxy instead of asterisk. i can also connect directly to my voicepulse connect account with the xten softphone and things work great. so i think i have the xten client configured properly and i know that the sveasoft firmware isn't throwing a monkey wrench into the picture. i suppose i could configure ser to "front" asterisk since it appears to deal with the nat, but i'm wondering if i'm missing something basic. my channel config files look like the following: sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=supersecret ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no ; Typically set to NO if behind NAT qualify=500 [2001] type=friend ; This device takes and makes calls username=2001 ; Username on device secret=supersecret2 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip-internal nat=yes canreinvite=no ; Typically set to NO if behind NAT qualify=500 iax.conf [general] port=5036 bandwidth=low disallow=all ; Disallow all codecs allow=ulaw ; Allow codecs in order of preference jitterbuffer=no [voicepulse] context = voicepulse-in secret=topsecrect auth=md5 type=friend host=gw5.voicepulse.com [1] http://www.google.com/search?q=retrans_pkt:+Maximum+retries+exceeded+on+call++site:http://lists.digium.com&hl=en&lr=&ie=UTF-8&start=10&sa=N [2] http://www.sveasoft.com/modules/phpBB2/index.php
I see this message in my asterisk log sometimes. Can someone explain to me what this means and how to correct the problem? May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 795fcf0c6 b267dda@10.0.0.6 for seqno 18950 (Non-critical Response) May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 0ec0538d9 ae6ac0c@10.0.0.6 for seqno 40146 (Non-critical Response) May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 0ec0538d9 ae6ac0c@10.0.0.6 for seqno 40147 (Non-critical Response) Thanks
None of our phones are being forwarded unless the phones are being forwarded unknowingly. >We see this when person A forwards their phone to person B, who has >forwarded their phone to Person A. >so A->B->A >or A->B->C->A and so on and so forth :) Joel Jn-Francois wrote: > I see this message in my asterisk log sometimes. Can someone explain to > me what this means and how to correct the problem? > > May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum > retries exceeded on call 795fcf0c6 > b267dda@10.0.0.6 for seqno 18950 (Non-critical Response) > May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum > retries exceeded on call 0ec0538d9 > ae6ac0c@10.0.0.6 for seqno 40146 (Non-critical Response) > May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum > retries exceeded on call 0ec0538d9 > ae6ac0c@10.0.0.6 for seqno 40147 (Non-critical Response) > > Thanks
> I see this message in my asterisk log sometimes. Can someone explain to me > what this means and how to correct the problem? > > May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries > exceeded on call 795fcf0c6 > b267dda@10.0.0.6 for seqno 18950 (Non-critical Response) > May 17 13:39:12 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries > exceeded on call 0ec0538d9 > ae6ac0c@10.0.0.6 for seqno 40146 (Non-critical Response) > May 17 13:39:19 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries > exceeded on call 0ec0538d9 > ae6ac0c@10.0.0.6 for seqno 40147 (Non-critical Response)We tend to get this when asterisk tries to call a SIP extension which has lost its connection for some reason (network troubles, power outage, whatever). See if there are any calls being attempted at that time... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett
Hi, I phone with phpagi and/or x-pro. Sometimes I get this warning in the asterisk-console: "maximum retries exceeded on call". I noticed when this message shows up, asterisk hangs up the call (even when i'am in the middle of a call, according to our employess) When they restart x-pro it seems to work properly again (at least some time). Asterisk and the clients are in the same LAN. I read the FAQ at voip-info.org but it didn't help. Here is my sip.conf -------------------------- [general] context=telin port=5060 bindaddr=0.0.0.0 srvlookup=yes toos=lowdelay allow=g726 allow=ulaw rtptimeout=60 rtpholdtimeout=300 useragent=EASYCOM nat=yes ------------------------- after that comes the whole register-thing here comes a sample user (all are the same) ----------------------------- [user] context=telout type=friend secret=XXX dtmfmode=rfc2833 host=dynamic allow=all canreinvite=no ----------------------------- in x-pro everything is standard (nothing changend but the network-settings and sip-proxy) Since Iam neither a linux nor a asterisk-crack, I don't really have a clue what's going on. Hope you can help me :) Regards Michael -- Immosky AG Service-Zentrale Dufourstr. 5 CH-8702 Zollikon-Z?rich Tel +41 (0)43 344 52 52 Fax +41 (0)43 344 52 58 www.immosky.ch haeberle@immosky.ch