similar to: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy

I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (the 90xxxxxxxxxxx is a valid number, changed to protect the guilty): == Spawn extension (from-sip,

Hi I wonder if anyone can throw some light on the * console message. This only occurs when I register a phone on the end of a BT ADSL line, with a Draytec router/modem. The phone registers okay but cannot dial out. Console message: Notice[1125329600]: File chan_sip.c, Line 1759 (copy_via_headers): No field 'Via' present to copy Thanks Steven *****************************************

People: Did you have seen this message before? noc2pbx*CLI> -- Executing Goto("SIP/-081363a0", "ivr1|2000|1") in new stack -- Goto (ivr1,2000,1) -- Executing Answer("SIP/-081363a0", "") in new stack -- Executing Wait("SIP/-081363a0", "1") in new stack -- Executing BackGround("SIP/-081363a0",

Szenario: UA(Grandstream) => PROXY(SER) => GATEWAY(*) => PSTN After sending the SIP ACK From Gateway (*) ACK sip:123456@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5 From: "Me" <sip:123456@mydomain.de>;tag=0f63d269bc25545d To: <sip:100@mydomain.de>;tag=as05df60b5 Contact: <sip:100@192.168.0.1> Call-ID:

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

Hi all: I've no response for the last question with the same subject. Please excuse me for the extreme length of this mail, but I send 2 SIP traces. I have problem with * and 5300, when the incoming and outgoing call are routed thru the same SIP gateway (AS5300). Do I need to set an special things in sip.conf? First all, the * printout. Second, the 5300 trace. Thanks in advace, Gus

--? qq:651112986???????????]??????158-0025-9602 ???sihi ???????? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.alioth.debian.org/pipermail/pkg-xen-devel/attachments/20131220/e3c58816/attachment.html>

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060 SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:

In case this is relevant: kernel: [17549.714369] imap-login[11704]: segfault at 0 ip b75c3d23 sp bf888178 error 4 in libc-2.11.2.so[b7551000+140000] If no host field attribute is present in ldap (like mailHost for qmail-ldap), and we are trying to use proxy, then dovecot segfaults. If more information's needed, let me know. Also, somewhat unrelated, but maybe the wiki needs to be edited

I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that is behind a network device to which I don't have ready access, which is performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 system on a public IP. My question is very specific, and I don't need right now to discuss the ins and outs of the above setup. What I am seeing is that when I have set up a

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Hi All, I'm running trixbox 2.0. The problem: a remote extension behind a NAT, can call other extensions, can call any other party, can call voicemail, will ring when rung, but when answered there is nothing and the dialling party continues to hear the ring tone. I'm getting this error in the logs: "That's odd... Got a response on a call we dont know about" I see

Hi, I''m trying to find a way to configure the firewall on centos systems using puppet by copying the /etc/sysconfig/iptables file and restarting the iptables service. That part is simple to accomplish. The twist I need to add though is that I only want to do this if a copy of that file for the node exists on the server. If this file doesn''t exists then I want puppet to not

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hi, the attached patch is very important for Cygwin. I don't know how I could have missed that for months now :-( I hope this can be included in 2.5.1p2. The pw_gecos field in Cygwin's /etc/passwd contains Windows specific authentication informations which let NT domain users logon to a machine without the need to inform the logon server (sshd in our case) about the name of the NT

Hi all, I can get access to the data stored in globals as follows: GlobalVariable* pGvarAct=_set_a_valid_global_; const_ptr_indicesVec=_the_indexes_ llvm::Constant* pConst; pConst=ConstantExpr::getGetElementPtr(pGvarAct, const_ptr_indicesVec); and e.g. use this to repalece an argumrnt of an instruction like: I->setOperand(someArgumentIndex,pConst); However, getGetElementPtr returns a

Dear all I am trying to find the best way to handle large fields in reference classes. As the code below shows assignment via <<- causes many copies to be made if the subsetting is extensive (in modb1). This can cause R to run out of memory. Creating a local copy and using the optimisation in <- is the best solution I have found so far (in modb2) - but it is not really much better than

Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). Nortel did an upgrade which changed a bunch of things today, so I thought I'd give it another shot. It looks like I'm much closer this time, but still no go. Can't do calling in either direction. Anyone have any ideas? Thanks! Shawn [nortel] host=10.0.0.10 insecure=very type=peer qualify=no