similar to: Pedantic=yes not working

Hi guys i'm using asterisk 11.18.0. I need to send the pound # sign to my SIP provider, but each time it's reencoded in %23. I try to put pedantic=yes in the sip.conf as advised here: http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic but nothing's changed. Have someone already met this issue please ? thanks a lot, regards, Alan

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

Hello developers. I have made a yes/no 'pedantic' patch that affects how ssh command line and other agents that employ yes/no prompts interpret results. The nature of the patch is the following: When prompted with things like: ----SNIP---- ./ssh localhost The authenticity of host 'localhost (127.0.0.1)' can't be established. RSA key fingerprint is

Hi, I'd like to use the compiler flags -ansi -pedantic -Wno-long-long in my project but some llvm headers cause errors with them. I tried to compile llvm with these flags but it proved to be quite difficult. The headers, however, seemed easier to fix. Almost all of the errors were commas at the ends of enum lists or extraneous semicolons after namespaces. I went ahead and hunted down

G++ 4.x.x with -pedantic option reject comma at end of enumerator list: Attached patches let LLVM Instruction.h and Type.h headers compile using G++ 4.x.x with lot warrnigs but without errors. Vladimir -------------- next part -------------- A non-text attachment was scrubbed... Name: Instruction.h.patch Type: application/octet-stream Size: 1332 bytes Desc: not available URL:

Chris said something about Xcode fixing this ? It doesn't happen with the GCC 3.4.6 compiler on Linux so that's why I missed it, but I thought Chris had fixed it. Reid. On Fri, 2006-11-03 at 08:48 -0400, Jim Laskey wrote: > The build is failing with; > > LiveIntervalAnalysis.cpp:218: error: floating constant exceeds range > of 'double' >

The build is failing with; LiveIntervalAnalysis.cpp:218: error: floating constant exceeds range of 'double' LiveIntervalAnalysis.cpp:253: error: floating constant exceeds range of 'double' LiveIntervalAnalysis.cpp:328: error: floating constant exceeds range of 'double' LiveIntervalAnalysis.cpp:1350: error: floating constant exceeds range of 'double' If I

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

Thanks very much for the responses on my last three issues. I've incorporated the appropriate changes into the budding Debian package. I feel embarrassed to raise this one, but in the interests of shutting lintian up maybe this could be fixed at some point: I: hivex: spelling-error-in-manpage usr/share/man/man1/hivexregedit.1.gz reencode re-encode I: libhivex-perl: spelling-error-in-manpage

I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI> [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

I've setup Asterisk to work with Broadvoice for both incoming and outgoing calls. I can make outgoing calls, but when I try to receive an incoming call I see the following message on the console: [date] NOTICE[8661]: chan_sip.c:13178 handle_request_invite: Unable to create/find SIP channel for this INVITE It's registered with Broadvoice: Name/username Host Dyn

I'm experiencing a "Failed to authenticate on INVITE" error (see output below) whenever I try to MAKE a call through the Broadvoice account. I noticed some others had the same problem but it went away when they rebuilt Asteris w/ a new version. N such luck for me! I'd be grateful for any assitance. Here's what I've done so far: 1) I downloaded the latest stable

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap