similar to: retrans_pkt: 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- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

Searched the archives thoroughly... Can't find this specific problem... Simple setup with Asterisk on RedHat. No voice cards in the box, 2 SNOM 200 phones... Phones seem to work well, can leave VM, Message Waiting Indicator lights up but when I try to retrieve messages the call terminates and the following happens: -- Executing VoiceMailMain("SIP/520-a25e",

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 5921 bytes Desc: image001.jpg Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/74245e28/attachment.jpeg

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 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

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

Hi, i've just got 2 grandstream phones and when I try to connect them with * I get the following: -- Playing 'demo-abouttotry' (language 'en') WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for seqno 59134 (Response) I've seen there was some discussion on this already but i

First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:10000-20000 10000-20000

Hi, I just updated my image from CVS, compiled and reinstalled it. Now whenever I make calls from my Grandstream phone to my X-Lite soft-phone, the call does not complete correctly. Scenario: 1. I take the GS off hook and dial 1100 (the extension of the x-lite phone). 2. The x-lite phone rings properly. 3. The user at the x-lite site answers the call. 4. The GS phone continues to

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, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >

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 there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

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

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

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

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

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