RE: [asterisk-users] Bad Echo between SIP calls

Best way to do this is not touch the sip.cfg, ever. Leave it as included in each release and include your overrides in a different file. Then reference your files like this in your MAC.cfg file, your file will override the sip.cfg defaults. CONFIG_FILES="phone_user.cfg,server.cfg,sip.cfg" In se

Re: [asterisk-users] Hot GXP-2000

On 6/10/07, Bill Hackensack <[EMAIL PROTECTED]> wrote: Why? Because of their excellent customer support in taking care of a problem? At the price of the Grandstreams compared to others, I can deal with a couple of bad apples. I can buy two Grandstream's for the price of a phone with similar fe

Re: [asterisk-users] Bad Echo between SIP calls

It doesn't matter if it's supported, they are all, however I have seen some echo problems after firmware upgrades, the only way to fix it was to either copy the differences or overwrite my old config files with the new ones that came with the firmware and then modify as needed for my setup. On 6/

Re: [asterisk-users] Bad Echo between SIP calls

The sip config & firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP-->SIP & wouldnt mind to share the sip.cfg, sip.ld & bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I ca

Re: [asterisk-users] Noise on FXS ports (Sangoma)

John Novack wrote: > In this troubleshooting case, it probably is better that there is NO > dialtone, which would make the hiss easier to hear. > I am curious what the OP found > When Asterisk is stopped, does the hiss continue? Okay -- I have tested this and yes, the hiss is still present even af

Re: [asterisk-users] Hot GXP-2000

Why? Because of their excellent customer support in taking care of a problem? At the price of the Grandstreams compared to others, I can deal with a couple of bad apples. I can buy two Grandstream's for the price of a phone with similar features. I can deal with a lot of bad apples at that rat

[asterisk-users] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??

I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86d

Re: [asterisk-users] Bad Echo between SIP calls

Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood <[EMAIL PROTECTED]> wrote: Stephen Davies wrote: > On 09/06/07, Deepak Naidu <[EMAIL PROTECTED]> wrote: >> Ya, I have

Re: [asterisk-users] Polycom 301 vs. 330

Jeff Davis wrote: Lee Jenkins wrote: Is the main difference between the two the full duplex speaker on the 330? The 330 also includes two other features over the 301: - A 103x33 pixel graphical display as opposed to the 4 line x 20 char. monochrome on the 301. - Built-in 802.3af PoE. It

[asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event

Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the foll

Re: [asterisk-users] Polycom 301 vs. 330

Lee Jenkins wrote: Is the main difference between the two the full duplex speaker on the 330? The 330 also includes two other features over the 301: - A 103x33 pixel graphical display as opposed to the 4 line x 20 char. monochrome on the 301. - Built-in 802.3af PoE. It does not support Cis

[asterisk-users] Polycom 301 vs. 330

Is the main difference between the two the full duplex speaker on the 330? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.c

Re: [asterisk-users] OT: CallManager ANI restamp.

On Sun, 10 Jun 2007, Mattt wrote: Try http://cisco.com - pretty certain they support their own proprietary, very expensive kit ;-) If Cisco had solutions for me, I wouldn't be humiliating myself asking this question on this forum. -- Alex Balashov Evariste Systems Web: http://www.eva

Re: [asterisk-users] IAX trunk with dynamic IPs

Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set "enable=yes" in the "[general]" section of /etc/asterisk/dnsmgr.conf* --

Re: [asterisk-users] OT: CallManager ANI restamp.

Alex, While we're at it, I have this Mitsubishi Magna with a computer-controlled transmission. After a voltage sag, the TCU seems to have "gone to lunch". Forcing it into limp mode by pulling the TCU fuse seems to right things (although it's then duly stuck in third gear). I know this has not

[asterisk-users] OT: CallManager ANI restamp.

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new le

Re: [asterisk-users] IAX trunk with dynamic IPs

*set "enable=yes" in the "[general]" section of /etc/asterisk/dnsmgr.conf* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asteris

Re: [asterisk-users] IAX trunk with dynamic IPs

On 6/9/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote: First of all, thanks for your help. I just want to check if I understood. If a set the TTL for 10 seconds for host.no-ip.org and configure the parameter host as "host=host.no-ip.org", Asterisk will try to find the IP address of host.no-ip.or

Re: [asterisk-users] IAX trunk with dynamic IPs

Hi Noah, First of all, thanks for your help. I just want to check if I understood. If a set the TTL for 10 seconds for host.no-ip.org and configure the parameter host as "host=host.no-ip.org", Asterisk will try to find the IP address of host.no-ip.org each 10 seconds? That is it? Thanks again

[asterisk-users] remove

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, June 09, 2007 5:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP Hi. Cal

[asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP

Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see tha

[asterisk-users] H.323 trunk between MD110 and Asterisk

Hi. Anyhone have any experience with trunking between Ericsson MD110 and Asterisk using H.323? I've tried both ooh323 and the /channels/h323 one in version 1.4.4 and 1.4.0 of Asterisk. ooh323 does not manage to establish the call (starts to ring but then disconnection when answering the cal

Re: [asterisk-users] Hot GXP-2000

One of the reasons why I stand clear of Grandstream - Original Message - From: "Carlos Chavez" <[EMAIL PROTECTED]> To: "Asterisk" Sent: Friday, June 08, 2007 6:47 PM Subject: [asterisk-users] Hot GXP-2000 ___ --Bandwidth and Colocation

Re: [asterisk-users] Bad Echo between SIP calls

Stephen Davies wrote: On 09/06/07, Deepak Naidu <[EMAIL PROTECTED]> wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. & SIP cals were great on them. & now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting ec

RE: [asterisk-users] Bad Echo between SIP calls

reinvite is disabled. Also its a Dell PowerEdge 850 server running asterisk connected to a Cisco switch. & other network in company have Cisco Switch. Also we have approx 75 Polycoms all over. canreinvite=no -- Deepak Steve Totaro <[EMAIL PROTECTED]> wrote: v\:* {beha

Re: [asterisk-users] Bad Echo between SIP calls

Yeah I have made sure its the correct port. We have 75 polycoms currently. ? the SIP-to-SIP echo is there. -- Deepak "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> wrote: Deepak Naidu wrote: > Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to > PRI we we

[asterisk-users] Re: SIP & NAT ...

On Fri, 1 Jun 2007, Gordon Henderson wrote: So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct

Re: [asterisk-users] No sound, problem is not a NAT

Hi Carlos - HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However,

Re: [asterisk-users] No sound, problem is not a NAT

What about the RTP ports (rtp.conf). Aren't they blocked on your firewall/iptables? Rgds, Ricardo Martins. Carlos Jerónimo escreveu: HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asteri

Re: [asterisk-users] IAX2 Trunk No Sound

> 3. Can you post some of the CLI errors you mentioned? > > iax2_trunk_queue: Maximum data space exceeded > > and once this start it never gets stopped so I've to kill the > asterisk and restart the whole box. Instead of restart whole box if > I just try to restart the asterisk my agents not able

[asterisk-users] Linksys 941/942 reboot and persistent MWI

We've got a bunch of Linksys 941/942s and have them all configured to upgrade the config periodically. Problem is, when the phone loads a new config it goes through what appears to be a soft reboot, although it only takes about 5 seconds. During this time, the display goes blank and the (normally

[asterisk-users] No sound, problem is not a NAT

HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound come

[asterisk-users] R2 Argentina

Dear Folks, I have found that Argentine variant ar libmfcr2.0.0.3 is not set correctly Regarding ANI restriction signal. Argentine regulations since 1999 have swaped SIG_12 with SIG_15 in order To restrict ANI presentation to the user. I dont know if it has been patched in later releases of mfcr2

[asterisk-users] Is There any Asterisk TODO(Developer side) list Available

Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --Bandwi

RE: [asterisk-users] Bridged PRI calls - processor involvement?

Are you running recording on your box or FTPing large recording files or PDFs or anything other than just voice traffic? Has voice traffic spiked in conjunction with your problems? Are you doing any kind of port monitoring/mirroring on your switch? Most people look at the 100mb or 1Gb figure but

RE: [asterisk-users] Bad Echo between SIP calls

Do you have reinvites enabled? Are you running this over a linksys four port SoHo router/switch or something? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMA

RE: [asterisk-users] Bad Echo between SIP calls

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Stephen Davies > Sent: Saturday, June 09, 2007 4:00 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Bad Echo between SIP calls > > On 0

Re: [asterisk-users] Bad Echo between SIP calls

Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium & no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before

Re: [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity

On 08/06/07, Asterisk <[EMAIL PROTECTED]> wrote: Would a good 1 gBit switch be enough to handle that (Asterisk box would be connected to that switch with 1 gBit connection, and computers with Microsoft RTC Library would be connected with a 100 mBit connection)? Alex: 30 concurrent calls will b

[asterisk-users] Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone

Oops, I had some problems and was offline unable to remind you about the conference yesterday. LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash player, will autostart) THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the new book, whick looks like a great effort.

Re: [asterisk-users] Bad Echo between SIP calls

On 09/06/07, Deepak Naidu <[EMAIL PROTECTED]> wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. & SIP cals were great on them. & now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP cal

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

Alvaro... Hum..., I never have tried RxFax... let me know if you need any extra help with that. Sounds interesting On 6/8/07, Alvaro Parres <[EMAIL PROTECTED]> wrote: Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke ev

Re: [asterisk-users] Asterisk 1.4 with Unicall

Hi Carlos, On 6/8/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the que