[Asterisk-Users] Maximum retries exceeded on call/phantom calls?
I am confused due a side effect produced in my * installation. It consists of 1 Sangoma A101 E1/lSDN PRI card connected to Telmex service 16 analog phones thru SIP enabled SP5004 Micronet gateways 4 SIP hard phones. Everything in a local network/no natting. We are processing nearly 2000 calls/day outgoing/incoming Everithing seems to be ok but after an hour or so I begin to see the message Maximum retries exceeded on call... On my logging console. This message continues to appear with a climbing frequency on different call ids till the entire system begin to unregister my sip clients. Asterisk needs to be restarted as if it has suffered a DOS attack. Prior to this situation arrives, I notice that phantom calls rings phones but nobody there--- After a couple of weeks of debugging I notice that this situation could be related to 3-way calling from the operator to other sip extensions. This tranferred calls seems not to die after the normal operation of the feature (flash/get tone/dial extension/speak with employee/hangup). I have all my sip gateways set to support transfer, so SIP attended transfer is done by the gateway and by zapata at the same time producing the side effect? Waiting some feedback OCA -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 267.15.2/251 - Release Date: 04/02/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call.
Using SIP? IAX? One way sound is usually a SIP and nat/firewall problem, make sure ports are forwarded. Steve - Original Message - From: Peter Ankerstål [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 10:39 PM Subject: [Asterisk-Users] Maximum retries exceeded on call. I have set up a asterisk-server behind NAT and peers to another asterisk and uses that one for outgoing calls. I have som clients on my asterisk and they could register to it well over internet. Not a problem. But when they try to call me the asterisk-server tells me this: Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response) Configs can be found at http://www.pulia.nu/~peter/asterisk/ When they call me they can hear me but I get no sound. Weird. Any Ideas? -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call.
Yes, using sip. The ports are forwarded. The calls going to the other asterisk server works fine. The problem occurs only when people who are registred to my server tries to call. On Thu, 13 Oct 2005 08:30:17 +0100 Steve Daniels [EMAIL PROTECTED] wrote: Using SIP? IAX? One way sound is usually a SIP and nat/firewall problem, make sure ports are forwarded. Steve - Original Message - From: Peter Ankerstål [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 10:39 PM Subject: [Asterisk-Users] Maximum retries exceeded on call. I have set up a asterisk-server behind NAT and peers to another asterisk and uses that one for outgoing calls. I have som clients on my asterisk and they could register to it well over internet. Not a problem. But when they try to call me the asterisk-server tells me this: Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response) Configs can be found at http://www.pulia.nu/~peter/asterisk/ When they call me they can hear me but I get no sound. Weird. Any Ideas? -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call.
I have set up a asterisk-server behind NAT and peers to another asterisk and uses that one for outgoing calls. I have som clients on my asterisk and they could register to it well over internet. Not a problem. But when they try to call me the asterisk-server tells me this: Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response) Configs can be found at http://www.pulia.nu/~peter/asterisk/ When they call me they can hear me but I get no sound. Weird. Any Ideas? -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] maximum retries exceeded on call
has somebody an advise. Do I need to provide more information? Regards Michael Michael Häberle wrote: 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-Zürich Tel+41 (0)43 344 52 52 Fax +41 (0)43 344 52 58 www.immosky.ch [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] maximum retries exceeded on call
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-Zürich Tel+41 (0)43 344 52 52 Fax +41 (0)43 344 52 58 www.immosky.ch [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 40147 (Non-critical Response) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
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 :) No matter how many times I tell 'em ... Julian 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 40147 (Non-critical Response) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] for seqno 40147 (Non-critical Response) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
i am server with ser(iptel) and asterisk. the prosess is: the xlite the connect to ser, the ser redirect to asterisk and call for x100p after the two minutes the call, the call is cut. the error is: Asterisk Ready. *CLI -- Executing Dial(SIP/sorcier.com.pe-41000490, Zap/1/499732) in new stack -- Called 1/499732 -- Zap/1-1 answered SIP/sorcier.com.pe-41000490 -- Hungup 'Zap/1-1' == Spawn extension (incoming_ser_asterisk, 0499732, 1) exited non-zero on 'SIP/sorcier.com.pe-41000490' Apr 11 01:21:09 WARNING[1897]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 55926 (Critical Response) wath happend? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
Hi: I have a asterisk server that shows the following Warning Jan 24 10:23:37 WARNING[1116941120]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Reque If it show on the midel of the call rhe call will be droped or if it show at the begining of the call the call will show buisy ( No one is available to answer at this time) Any help will be appreciate Please help Thanks Erick W ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
Any idea why I keep getting this on my Asterisk console? This is coming from a Grandstream 101.. No NAT or anything, clean connection between phone and Asterisk on same subnet via 100mbit. *** Oct 26 15:19:52 WARNING[180235]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
Eric C. Snowdeal III wrote: 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 [EMAIL PROTECTED] for seqno 41270 prompted by a recent email to the group [1] about setting the bindaddr, i took a closer look at the sip messages being sent back and forth and noticed that the contact header was incorrectly set to 127.0.0.1 in the 200 o.k. message [2]. once i set the bindaddr to the * machine's public ip address everything worked fine and and contact header i.p. address was set correctly. what's odd, at least to me, is that unlike the recent email about a similar issue [1], my * box is on a non-natted, public ip address so i would have thought that keeping the default bindaddr (0.0.0.0) would have worked, but obviously it didn't. not sure how to interpret the dirth of responses, perhaps this was frighteningly obvious to everyone else. [1] http://lists.digium.com/pipermail/asterisk-users/2004-June/051375.html [2] RECEIVE TIME: 7548279 RECEIVE my.public.asterisk.ip:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip:5060;rport;branch=z9hG4bKFE76D502C2BF11D8A741000D93C22AF4;received=my.wan.ip;rport=1029 From: snowdeal sip:[EMAIL PROTECTED];tag=1666554831 To: snowdeal sip:[EMAIL PROTECTED];tag=as7f7ed33f Call-ID: [EMAIL PROTECTED] CSeq: 43970 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 1800 Contact: sip:[EMAIL PROTECTED];expires=1800 Date: Sun, 20 Jun 2004 13:44:34 GMT Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
i'm new to asterisk and am having trouble placing outbound calls. i Bug Grandstream so that they finally fix their buggy software. The GS phone sends occassional SIP packets to port 0, not to port 5060, as tcpdump or (better) ethereal will show you. There's a page on this at voip-info.org. I'd love to see that we e-mail in MASSES to Grandstream, so that they fix their software. The problem is that it doesn't happen always. Try [EMAIL PROTECTED] :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
Holger Schurig wrote: i'm new to asterisk and am having trouble placing outbound calls. i Bug Grandstream so that they finally fix their buggy software. The GS phone sends occassional SIP packets to port 0, not to port 5060, as tcpdump or (better) ethereal will show you. There's a page on this at voip-info.org. thanks for the heads-up about grandstream, but as i stated in the original message, i'm using xten lite softphones. hopefully this is the approproriate forum for this question; i believe this is not an xten configuration issue because i can connect to a ser/rtproxy/nathelper server without problems and i can connect directly to a voicepulse account, which leads me to believe that this is an * configuration problem on my part. less likely, i suppose, is the chance that * isn't as robust in handling nat than ser or whatever voicepulse is running. given the configuration files that i posted in the original message, are there any changes that i should make? certainly the asterisk faq makes the solution seems straighforward [1]: Most likely you have a SIP client behind NAT that is trying to communicate with Asterisk without having the nat=yes setting in place in sip.conf. Another cause for this could be related to a user device that has an sip entry but has been physically removed (switched off or LAN-disconnected). but as my original message showed, i do have nat=yes in my sip.conf and i don't believe the latter scenario is true. any help is greatly appreciated. [1] http://www.voip-info.org/wiki-Asterisk+FAQ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 [EMAIL PROTECTED] 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.comhl=enlr=ie=UTF-8start=10sa=N [2] http://www.sveasoft.com/modules/phpBB2/index.php ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries exceeded on call
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- [EMAIL PROTECTED] 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- [EMAIL PROTECTED] for seqno 102 (Request) Mar 17 16:43:43 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 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 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
check NAT setting try taking it out of sip.conf, that worked for me Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
On Tue, Mar 16, 2004, Barry Fawthrop thus spake: check NAT setting try taking it out of sip.conf, that worked for me Nope. My bad-shoulda said up front that I've tried both with and without nat=yes in sip.conf, no difference in symptoms. Regards, Ed Hintz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded on call
check context perhaps try include in the extensions.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
