similar to: Asterisk 1.6.0.6 sip doesn't work?

Hi all!! I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk 1.4.18. Both are home PBX's and both boxes register to a SIP DID at exactly same provider. One box runs without errors on the console, the other box keeps repeating : [Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet When i set debug on, it seems to

Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do : cd /usr/src/dahdi-linux-2.1.0.4 make clean ; make all ; make install cd /usr/src/dahdi-tools-2.1.0.2 ./configure ; make clean ; make all ; make install ; make config So far.... so good but then when i do : cd /usr/src/asterisk-1.6.0.6 make clean ; ./configure ; make all ; make install i

Hello, I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly with a CISCO mediaGW in order to send faxes to the PSTN using T.38. When Asterisk sends the initial INVITE containing the T.38 media offer in the SDP, the CISCO answers with a 488 Not Acceptable Media. Apparently, it looks like a configuration problem in the CISCO, but I have tested the CISCO with the Zoiper

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

Hi all I have been hacked but no idea how!!! I noticed somebody in Eastern Europe came from an American IP and tried to call loads of international numbers. Thankfully I had no credit with my VOIP out provider so the calls went nowhere. But if I had credit it would all have been used up. I noticed hundreds of calls being made from clid and src being either UNKNOWN or as ASTERISK. Here are a

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as "maxBitRate". Without capital "M". Are you closer to

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

Just a warning for you all that are using Nokia series E phones for SIP function. I updated my phones firmware today using the Nokia Updater, and now the SIP functionality, which previously worked pretty well is completely broken. The phone no longer registers with asterisk, although it displays the little icon as though it has, and it doesn't even seem to try to pass calls to

Hi, I am trying to use asterisk open source version(asterisk-1.6.0.5) with MySQL (using res_odbc)support for extensions and users list. The call rate is 7 calls / second and each call stays for 120 seconds. after making 25000 successful calls , calls started failing with following message on CLI. [Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable to create channel of

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

Hi! I noticed when upgrading asterisk that the latest version of asterisk is not recognizing the version of asterisk-addons properly. When you clean out /usr/lib/asterisk/modules and then install zaptel-1.2.12 -> libpri-1.2.4 -> asterisk 1.2.14 -> asterisk-addons-1.2.5 and then you compile and install asterisk *again* it complains that the modules of asterisk-addons are not built

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

we tested asterisk 1.6.2.0, found that when call from one sip_channel to another sip_channel , ------------------------------------------------------------------ exten => _X.,1,Noop() exten => _X.,n,Dial(SIP/${EXTEN},50,TtgM) ------------------------------------------------------------------ in asterisk 1.6.2.0 ,when sip user config to use dtmfmode=rfc2833 , it's ok, but when both