search for: sonus

On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein <damin at nacs.net> wrote: > Hello, > Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? > We are unable to get reliable RFC 2833 DTMF working, and have instead had to > use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this issue? > Welcome to the club! ;) I'll be blogging about this late...

When we make calls out of asterisk to the PSTN via a SIP termination service provider the called party gets a slight echo of their voice. Here is the setup; analog phone <> Linksys ata <> asterisk <> sip provider sonus GSX 9000 <> PSTN <> called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight echo of their voice. All of the Zapata.conf echotraining, echocancel, etc do not seem to apply here as there is no zap channel involved i...

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are canreinvite=yes and the PSTN termination path is canreinvite=no then calls between polycoms should not have asterisk in the media stream and wan link utilization is reduced. The problem look...

...all can be transferred to any of our other servers depending where the extension or queue reside. We would like to completely move the call off of the first box so we don't tie up resources on it. In our lab we are testing with 1.4.22.1 Our provider which delivers inbound calls to us uses a Sonus gateway. So far, testing has shown that if we transfer the inbound call prior to any media playback, it works. But, if the IVR plays media, then it is failing, with a 500 internal server error being returned. Thanks for any help -------------- next part -------------- An HTML attachment...

I recently got Festival performing Text to Speech on my Asterisk system. It is working great when I call from extension to extension in the house. But when I dial in on my phone number (which comes in on a sip registration to a Sonus server), I can not hear any sound. The asterisk box thinks it is playing the festival sound but I hear nothing. Any ideas?

Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But now when I do that I have no audio with call established. I think it's a problem of me not assigning the media IPs. How can I a...

Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several "invalid extension" or "password incorrect" messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might

...hannel bank). I cannot seem to get the T100P to send any clock to the channel bank. I prefer that it use the same clock source as the TE410P, but it doesn't matter if it's not in sync just as long as it's there. The TE410P is configured 3x pri_cpe, 1x pri_net. The three cpe go to XO Sonus switch, the net to legacy PBX. Clock is received from telco, old PBX receives clock from zaptel card, everything's green there, but the other card, the T100P, seems to not send any timing at all, as verified by our T1 analyzer, and is persistently in red alarm. In fact, even if I stick a loop...

Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since

..., but the remainder fail miserably. Using an account/username with an empty password for the affected ports fixes the problem - so this is something with www-digest method (?). I've spent 2 weeks debugging this with addpac development team, and the same device authenticates flawlessly with Sonus Proxy Server, SNOM Proxy Server, LongBoard Proxy Server, Nortel Proxy so this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.p...

...xtension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw Voicemail is recorded in wav|ulaw so there should be native playback to g.711 UAs and the wav is for windows email attachments. Outbound termination is to a Sonus GSX media gateway, g.711u is the only allowed codec for that peer (disallow=all allow=ulaw) Sip friend (the extension) has disallow=all allow=ulaw The entire RTP path should never be anything but ulaw! What is slin and what does this NOTICE mean? I have seen this question asked before,...

...t; message, yet T.38 negotiation achieves t38state 5 (chan_sip.c: T38_ENABLED) and calls are successful. I've been comparing Asterisk debug from both systems as well as wireshark captures, but I can't figure out why Asterisk is not sending the Linksys ATA's IP address. Broadvox uses a Sonus switch and gateway with separate IP addresses for SIP and media. Affinity uses "Sippy" (?) with a common IP for SIP and media. I believe I've already covered all the possible configuration scenarios. I just can't get the right detail out of Asterisk to determine if this is an Ast...

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

The national guard and/or army routinely implements VoIP over wireless in situations where comm is lost, I did see an news release that the Guard started this project in the south the day after the disaster hit. The key is not the VoIP infrastructure, that is the easy part (one ss7 Sonus softswitch and a DS3!), the key is distributing IP over a wide area, which is best done on the quick with WiFI and WiMAX like wireless technologies that can cover even areas submerged in water. WiFi VoIP phones now have new value... I am sure there are applications that the Guard does not priorit...

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

...r ugly world out > there then why not throw the RFC out of the window and *not* reject an > invite with a 488? It sounds like an interesting option to add to > "10"/trunk. Better secure than compliant & sorry. Why not do a little > Microsoft Embrace & Extent? Like e.g. Sonus and Cisco do with their > interpretation of SIP. > > Regards, > Patrick > > > > ------------------------------ > > Message: 4 > Date: Sat, 23 Jul 2011 12:07:49 -0400 > From: Paul Belanger <pabelanger at digium.com> > Subject: Re: [asterisk-users] Securin...

We are having an issue with Asterisk 1.6.1 and the MWI turning on when a user doesn't have voicemail. We see random MWI lights come on and the phone indicates a random number of messages (its been anywhere from 1-14) when a server reload is done. I just checked one user, they have no messages old or new and the phone (Polycom IP330) indicates that they have 2 messages. The user will check for

All, Having problems terminating to a Nuvox PRI, the tech at Nuvox is saying Asterisk is transmitting in CCITT (aka ITU?) when they're expecting (and will only accept) ANSI. The question is, is there a simple way to change this or am I stuck with rewriting code? I googled and checked the mailing list and found nothing, I could be barking up the wrong tree I guess. PRI is not my forte.

Hi, I'm just considering buying two Telecoms grade Sun Netra's to run a lab-based VoIP solution. Not my immediate thoughts as a VoIP platform, but from what I've heard, they can run Linux, and run it well. Only thing is: The Wiki and the Whitepaper just state that Asterisk is for the x86 architecture, but has been compiled to run on PPC architectures. No mention of UltraSparc.