similar to: Critical Transaction failed: Client non-INVITE - SJPHONE connected to Asterisk

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta

Hi all, I would like to hear from you, SjPhone has the option to Do not Send silence (audio options, advanced), should i use this or remove this option? Everything ran well until now, but there was few people on my server, i'm increasing sip extensions and i want to avoid complains from the users:) Best regards, Marco Mouta

Hi all, I've a some users on my network, reporting this: Sjphone is registered , and some times just looses registry in Asterisk, I don't know if it is expiration ( instead of loosing registry). Then to get registered again they need to restart their own PC. Why could this beeing happening? Best regards, Marco Mouta

[macro-hangupcall] exten => s,1,ResetCDR(w) exten => s,2,NoCDR() exten => s,3,Wait(5) exten => s,4,Hangup Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that Ast@home is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

On 11.17.1: The cli and the log are full of these warnings: WARNING[12110]: chan_sip.c:4086 retrans_pkt: Timeout on 849421411 on non-critical invite transaction. The number is a random 9-10 digits. What causes them? How do I stop them ? sean

Hi All I'm getting tens of thousands of these messages ever hour in the Asterisk CLI for Asterisk 13.22.0: [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 1924200000-502043860-301870737 on non-critical invite transaction. [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 301794058-652332923-1834701069 on non-critical invite transaction. [Apr

--- "Keith O'Brien" <keith@voipreviews.com> wrote: [phone1] type=friend username=keith secret=keith host=dynamic qualify=2000 disallow=g729 auth=md5 context=sip mailbox=9999 callerid="keith@10.1.1.12" <1000> But the log in SJPhone indicates that the registration is being rejected: 2003-09-25 18:55:34.776 UDP LOCAL->10.1.1.12:5060 REGISTER sip:10.1.1.12

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hello, I'm trying to register some dynamic SIP softphone (SJphone) but it's not very clear to me where (sip.conf??) I have to configure the registration. Can you give me an example of configuration file to register a dynamic SIP phone? Thanks Marco Poet QiNet SRL Via Oulx 30/A 10139 Torino Italy Email marco.poet@qinet.it Phone +39 0117410856 Fax +39 0117571140 Mobile +39 3493009702

Hi, I'm installing asterisk 1.0.5 for a partner in China. Since the ping round trip takes typically 600 msec, I doubt, whether voice quality will we satisfying, but that is currently not my concern. The problem is, that most SIP phones or software (e.g. SJPhone) do resend the invite request, after approx 500 msec (measured by ethereal). chan_sip from asterisk seems to have a special

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

Hi all, As i have already posted, i'm noticing a strange problem on my * server: It seems to me Asterisk is simply ignoring some of invites sent from my xlite 3.0. If i dial 2XXX numbers, all ok. If i dial 4XXX numbers that aren't accounts on asterisk i get answer from asterisk. If i dial 4XXX numbers that exist on my server nothing happens and i get call failed: Request timeout.

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;