[Sip-implementors] Please Help With This BYE

Hi All, I am trying to send BYE and the server sends 481 Call Leg/Transaction Does Not Exist. I am debugging my client with x-lite on another device. When the x-lite client sends bye it comes fine. Below is the ACK and the BYE that my client send. Thanks for any help, Nir ACK sip:1000@5.102.205.

Re: [Sip-implementors] sip over TCP

I would wireshark some client. On Mon, May 20, 2013 at 10:49 PM, Milton Sanchez wrote: > Hello, can someone please point me to the right direccion on finding some > infomarction/tutorial about SIP over TCP. > > regards > Milton > ___ > Sip-implementors

Re: [Sip-implementors] issue in sending ACK

Hi, Everything will START fine but B should BYE after 30 sec. Nahum On Sat, Mar 16, 2013 at 10:11 PM, satya r wrote: > Hi All, > > Let A is sending invite to B then B will send 180 ringing ,200 OK but A > will not sending ACK so > Is voice path will establish in between A and B ? > so what wil

Re: [Sip-implementors] Reason REGISTER & INVITE message faillure

You can INVITE without REGISTER so my guess is that if both fail and assuming that if the server is ok something is wrong with your account settings. On Fri, Feb 22, 2013 at 12:29 PM, Jan Bollen wrote: > Hi, > > Well, you might try to capture the exchanged SIP traffic with > Wireshark or look

Re: [Sip-implementors] Certification Agency to test my SIP Stack

n few accounts in some termination providers and check your stack Nahum On Fri, Feb 22, 2013 at 2:00 PM, isshed wrote: > Hi Nahum, > > > Could you please provide the name? > > Thanks, > > > > On Fri, Feb 22, 2013 at 1:24 AM, Nahum Nir wrote: > >> I know that the

Re: [Sip-implementors] Certification Agency to test my SIP Stack

I know that there is one company in Israel (forgot the name) that do stress test to SIP servers maybe you can ask them. On Thu, Feb 21, 2013 at 9:10 PM, Praveena Ss wrote: > hi isshed, > > i don't think any labs/organizations do only sip stack testing...but you > can do testing with so many ope

Re: [Sip-implementors] RTP Padding

and proceed with the rest of the data, as normal packet; if p=1. > > > On Sun, Feb 17, 2013 at 12:04 AM, Nahum Nir wrote: > >> Hi Everyone, >> >> In case that the other client is not supporting RTP padding will my client >> be notified in any way? >>

[Sip-implementors] RTP Padding

Hi Everyone, In case that the other client is not supporting RTP padding will my client be notified in any way? 10xs, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-imp

[Sip-implementors] Non Ascii Dispaly Name

Hi All, I'm implementing a SIP stack and having some problems with non ascii display name. The problem is that it is stored as short* and it needs to be converted to char*. What should be the MSB? Is it covered in any RFC? Thanks, Nahum ___ S

[Sip-implementors] Display Name

Hi All, Are there SIP servers that supports UTF8 as display name? Thanks, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

[Sip-implementors] CSEQ update upon resending

Hi All, After A timer fires what should I do with the SCEQ (both in case of first invite and in case of invite after 401 challenge)? Thanks, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.co

[Sip-implementors] CSEQ and timer question

Hi All, After A timer fires what should I do with the SCEQ (both in case of first invite and in case of invite after 401 challenge)? Thanks, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.co

Re: [Sip-implementors] Call Transfer

Thanks -Original Message- From: Rockson Li (zhengyli) [mailto:zheng...@cisco.com] Sent: Tuesday, February 22, 2011 3:53 PM To: Vivek Talwar; Nahum Nir; sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Call Transfer rfc5589 -Original Message- From: sip

Re: [Sip-implementors] Call Transfer

Thanks Talwar -Original Message- From: Vivek Talwar [mailto:vivek.tal...@aricent.com] Sent: Tuesday, February 22, 2011 3:44 PM To: Nahum Nir; sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Call Transfer Hi Nahum, Refer 3gpp spec number 3GPP TS 24.429

[Sip-implementors] Call Transfer

Hi All, Can someone please direct to the RFC where call transfer is defined? Thanks, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

Re: [Sip-implementors] Question About Hold

Thanks Paul. I need it in pure c, multiple Oss. -Original Message- From: Paul Kyzivat [mailto:pkyzi...@cisco.com] Sent: Thursday, November 04, 2010 8:40 PM To: Nahum Nir Cc: sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implementors] Question About Hold On 11/4/2010 12:24

Re: [Sip-implementors] Question About Hold

f to building a new one. Thanks, Paul On 11/4/2010 12:01 PM, Worley, Dale R (Dale) wrote: > > From: sip-implementors-boun...@lists.cs.columbia.edu [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Nahum Nir [hello.shalo

Re: [Sip-implementors] Question About Hold

torola.com] Sent: Monday, October 04, 2010 6:38 PM To: Nahum Nir; sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Question About Hold Hi Check RFC 5359 (Session Initiation Protocol Service Examples) Sections 2.1, 2.2 and 2.3. http://tools.ietf.org/html/rfc5359 Rega

Re: [Sip-implementors] Question About Hold

Thanks -Original Message- From: Avasarala Ranjit-A20990 [mailto:ran...@motorola.com] Sent: Monday, October 04, 2010 6:38 PM To: Nahum Nir; sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Question About Hold Hi Check RFC 5359 (Session Initiation Protocol Service

[Sip-implementors] Question About Hold

Hi All, I'm trying to implement hold. After capturing packets while using x-lite I noticed that upon hold it send an INVITE. How does the other side know that it means hold? Thanks, Nahum ___ Sip-implementors mailing list Sip-implementors@lists.

Re: [Sip-implementors] Please Help With that ACK

Iñaki Baz Castillo - Thank you so much!!! -Original Message- From: Iñaki Baz Castillo [mailto:i...@aliax.net] Sent: Wednesday, April 21, 2010 7:04 PM To: Nahum Nir Cc: sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implementors] Please Help With that ACK 2010/4/21 Nahum Nir

Re: [Sip-implementors] Please Help With that ACK

sponse="44c072095c7cfaa350e5f4bb4be8579a", algorithm=md5 Content-Length: 0 -Original Message- From: Iñaki Baz Castillo [mailto:i...@aliax.net] Sent: Wednesday, April 21, 2010 6:29 PM To: Nahum Nir Cc: Brett Tate; sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implemento

Re: [Sip-implementors] Please Help With that ACK

né CSeq: 1001 ACK Digest username="1000", realm="asterisk", nonce="3b1caf29", uri="sip:1...@81.218.154.62", response="44c072095c7cfaa350e5f4bb4be8579a", algorithm=md5 Content-Length: 0 -Original Message- From: Iñaki Baz Castillo [mailto:i...@aliax.ne

[Sip-implementors] Please Help With that ACK

Hi All, My call is getting disconnected after 30 seconds so I guess that there is something wrong with my ack. Please take a look at the dialog below and tell me what is the problem. Thanks, Nahum INVITE sip:1...@81.218.154.62 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060;r

Re: [Sip-implementors] What is wrong with that regisster

Alok, Thank you very very much!!! -Original Message- From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Alok 2 Tiwari Sent: Tuesday, April 20, 2010 7:36 AM To: Nahum Nir; sip-implementors@lists.cs.columbia.edu Subject

Re: [Sip-implementors] What is wrong with that regisster

10 AM To: Nahum Nir Cc: sip-implementors@lists.cs.columbia.edu Subject: Re: [Sip-implementors] What is wrong with that regisster 2010/4/19 Nahum Nir : > Hi all, > > I'm sending a register request to my asterisk server and it returns me a 401 > unauthorized. But when I'm s

[Sip-implementors] What is wrong with that regisster

Hi all, I'm sending a register request to my asterisk server and it returns me a 401 unauthorized. But when I'm sending back a register with digest it's keep sending me 401 unauthorized. The server log is not reporting of any errors. Below there is my server output. Please Help. Thanks, Nahum

[Sip-implementors] Please Help With REGISTER

Hi all, I'm sending a register request to my asterisk server and it returns me a 401 unauthorized. But when I'm sending back a register with digest it's keep sending me 401 unauthorized. The server log is not reporting of any errors. Below there is my server output. Please Help. Thanks, Nahum

[Sip-implementors] Please explain to me the following SDP negotiation

Offer v=0 o=- 4 2 IN IP4 192.168.2.103 s= c=IN IP4 192.168.2.103 t=0 0 m=audio 34362 RTP/AVP 0 101 a=alt:1 1 : hlOLopIA MH2kNKJY 192.168.2.103 34362 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:E44A2C3559FC4E91BD382CEB4A5EAE0F Answer v=0 o=M

[Sip-implementors] What is wrong with the following INVITE???

Hi All, I'm trying to make an INVITE so that the incoming audio will be received on 89.139.126.116:44074. I get Session In Progress and OK but no incoming audio. What is wrong??? Thanks, Nahum INVITE sip:+972547864...@gw1.man1.theiptele.comSIP/2.0 Via: SIP/2.0/UDP 89.139.126.116:44074 ;rport;b

[Sip-implementors] What is worong with that sip message???

Hi All, I'm trying to make an INVITE so that the incoming audio will be received on 89.139.126.116:44074. I get Session In Progress and OK but no incoming audio. What is wrong??? Thanks, Nahum INVITE sip:+972547864...@gw1.man1.theiptele.comSIP/2.0 Via: SIP/2.0/UDP 89.139.126.116:44074 ;rport;b

[Sip-implementors] Please Advise Regarding that SIP INVITE

Hi All, I'm sending the follwing INVITE and receveing 183 SESSION IN PROGRESS and OK. I'm expecting to receive audio audio at 85.65.11.138:45000 but it is not coming.  Any thoughts? INVITE sip:+972777041...@gw1.man1.theiptele.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.2:1;rport;branch=z9hG4bkpj6