RE: [Asterisk-Users] OT: SIP aware firewalls?

I know that I can stay with m0n0. The question still stands; are there circumstances when something more is required? Would something be gained by such a migration. I would think the only real circumstances where true SIP-aware firewalls would be required would be in an environment where one

RE: [Asterisk-Users] GSM Gateway / Terminal for sale

Single port GSM Gateway support 900 / 1800 GSM mode with external antenna. Brand new unit and all of them will be tested before dispatch. Extremely easy to setup and can be used out of the box without any configuration. So should be good alternatively of phonecell or nokia pbx etc..

RE: [Asterisk-Users] OT: SIP aware firewalls?

Until now I've only used IAX2 to connect to ITSPs. I've been toying with a SIP connection to Gizmo Project, but not yet successfully. It brings to mind a question. At what point does it make sense to consider a SIP-aware firewall such as those from Ingate? You should be able to run SIP

[Asterisk-Users] SIP/IAX softphones for use in call centre environments

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with

RE: [Asterisk-Users] Stay away from Grandstream!

Are you saying that we just wasted our money with our recent purchase of Grandstream phones? The last thing I need is problems with a phone. Someone please confirm…are these phones unusable? It seems that different people are getting vastly different results. In my experience, the

RE: [Asterisk-Users] Stay away from Grandstream!

We've contact Grandstream support, but they cannot help. Now we want to send devices to Grandstream for repair but they on longer reply mail! This is where a good reseller is worth their weight in gold. Unless you're buying massive quantities of the things (in which case a failure of 2 is

RE: [Asterisk-Users] Blackberry SIM card

(Bear in mind my knowledge of Blackberry stuff is UK-based, not US-based. How different it is over there I don't know) To get a sim card you need service from T-Mobile. The Blackberry devices are no different from any other GSM cellphone, so you can get a SIM card from anyone, not specifically

RE: [Asterisk-Users] Handytone 486 Outbound problem

Asterisk -r makes no mention of any activity when this occurs so it seems that Asterisk is not even generating the busy signal. Is the Handytone capable of doing this and if so, why would it be? Make sure early dial is disabled in your HT486 config. I've never been able to get it working

[Asterisk-Users] Handling SIP clients behind NAT on a semi-dynamic IP

Greetings all, A couple of clients have recently decided they'd like extensions to their office PBXs at their homes, so they've duly been provided with preconfigured phones which register with the Asterisk server at their offices (public IPs) quite happily. Every 3-5 days it seems that these

RE: [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue

I know it's bad form to reply to one's own messages, but I should have added that both boxes in question are running 1.2. I was under the impression that many of the IAX jitter buffer issues had been resolved in 1.2? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is

[Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue

Greetings all, I've been doing some testing with a new client's voicemail system from my asterisk box at home. I'm using a 7960 (alaw/ulaw) then an IAX2 link between my server and the client's server. I've configured IAX settings at both ends as follows: jitterbuffer enabled, trunking enabled,

RE: [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue

Jitterbuffer is to blame. I feared as much. Thanks for the confirmation. 1) Is this a known issue with ILBC + Jitter Buffer? 2) Why does the same not occur from a SIP handset? 3) Is it safe or wise to disable the jitterbuffer? Thanks kindly. Regards, Chris -- C.M. Bagnall, Director,

RE: [Asterisk-Users] Small / embedded system recommendations

About the 4801, Kristian said: - No FXS ports - the Soekris doesn't have the means to provide ringing voltage for the card. Doesn't it use the 5V rail of a standard molex connector to generate ring voltage? Or does it use the 12V rail. If it's the former, I think you could probably use power

RE: [Asterisk-Users] asterisk in real estate developments

I was wondering if anyone has used asterisk in a real estate development project. I know someone that is developing a ~400 home project and thought asterisk might be a possible alternative to the phone company and a way to offer more service to buyers. How about deploying asterisk to

RE: [Asterisk-Users] bristuff use without BRI/PRI

Just a quick question. I am looking into bristuff for app_devstate to use with Snom phones. I don't have a BRI card installed on this server. Almost all the documentation I can find assumes that a card is being used. I have a number of boxes that don't have BRI cards but still have

RE: [Asterisk-Users] Small / embedded system recommendations

Would anyone have recommendations for a small or embedded system suitable for running Asterisk on? Ideally, we'd like two boxes: - One using compact flash, and is fanless, with rapid booting. - One with a hard disk for voicemail, call recording, etc. Preferably they would be capable of

RE: [Asterisk-Users] Connecting asterisk over consumer wifi network

So, I'd like to get some feedback on how it might work if we simply put a wireless access point at each workstation, and used the 4 port switch to connect to the PC + polycom handset. In my experience, wireless signals have a really poor range in elderly buildings - they're usually built of

RE: [Asterisk-Users] UK ISDN2e with DDI?

The ringtone on your Grandstreams is indeed set in the phone itself. I think they hold up to 4 ringtones (default, custom 1 2 3) which can be configured either per line or different rings on different caller ID. Grandstream have a freely available utility to convert PCM ringtones into the

[Asterisk-Users] Zaphfc as a timing source?

Hello all, I know the TDM cards (and I assume the TE cards) provide a timing source to be used for IAX trunking etc., but is it possible to use a BRI card running under zaphfc as a timing source, or should one run ztdummy as well? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director,

[Asterisk-Users] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2

Posted a couple of weeks ago. Would be most grateful if some kind soul shed light on this please? Hello all, Since upgrading a couple of our servers to 1.2 I've noticed problems when talking to users on 1.0.9 servers. The servers are connected via IAX2 with trunking and jitter buffer enabled

RE: [Asterisk-Users] Looking for advice on cell carrier's default Unavaliable message

I don't know where you're based, so I've no idea if this'll work for your users. If you're using GSM mobiles there are a load of (reasonably) standard vertical service codes to enable/disable call forwarding etc. depending on conditions. How about this: 1) set up a DID that's never answered in

[Asterisk-Users] Kernel upgrade causes ztdummy to refuse to run

Hello all, I recently upgraded the kernel on one of the phone servers I have at home (dual Xeon 2.4) from 2.6.11 to 2.6.14 in the usual way, copying the .config file across and building the new kernel. Now ztdummy is refusing to run, and gives the following errors in dmesg: ztdummy: Unknown

RE: [Asterisk-Users] Kernel upgrade causes ztdummy to refuse to run

you need to recompile zaptel drivers... they use the kernel headers to build and since you change the kernel headers by upgrading your kernel... time to recompile cd /usr/src/zaptel make clean; make install Sorry to say, I've already tried that after each kernel recompile with different

[Asterisk-Users] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2

Hello all, Since upgrading a couple of our servers to 1.2 I've noticed problems when talking to users on 1.0.9 servers. The servers are connected via IAX2 with trunking and jitter buffer enabled (jitter buffer on default settings). Reading through posts in the list archives, there are a number

RE: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

You can, but only in the US I believe. I've never found any deal less than £150 (UK). I was quoted £36 a couple of weeks ago by one of the Cisco resellers a google search provided me with, if that's any help. I can't remember the company name I'm afraid... Regards, Chris -- C.M. Bagnall,

RE: [Asterisk-Users] A question about transfering calls

I have a question about transfering calls. If I transfer a call to extension 4000 and nobody answers I want the call to be returned bak to me at extension 1000. How do I do that? Any help is apreciated! many thanks! Try something like this: macro internal (dialstring, fallback, timeout)

[Asterisk-Users] Asterisk 1.2 and Athlon64 platforms

Good evening all, Are there any folks out there running Asterisk on Athlon64 platforms with 64-bit operating systems? I have a couple of new asterisk servers to build up this week and I'm debating whether to order some Athlon64 CPUs and boards for them. I usually install Gentoo onto the boxes,

[Asterisk-Users] Camping-on-busy

Hello all, I'm trying to write a macro that'll handle blind SIP transfers nicely, since at present, blind transferring to a busy SIP extension will give the incoming caller busy tones. Hopefully this will be of use to others on the list once it's working correctly. Here's what I've got so far:

RE: [Asterisk-Users] Would DECT cordless phones work with Asterisk andVOIP?

I've just heard about DECT which is used for about 50 million phones in Europe and is just starting to appear in the US. I didn't realise they'd not been around in the US for long. I've had DECT phones for at least 5 years now... In my house, a Uniden 5.8 and Panasonic 2.4 cordless system

[Asterisk-Users] Really lightweight itemised billing

Good morning all, I'm trying to find an application that'll do really lightweight billing for Asterisk CDRs. On our asterisk servers deployed at people's offices, we have CDRs being logged to PostgreSQL, which can then be analysed by the staff at those offices using a PHP-based CDR analyser.

[Asterisk-Users] 7960 audio quality when calling remote asterisk box

Hello all, I've been doing some testing with the 7960s I have here calling into a remote asterisk box (1.0.9). Audio quality on the 7960 is perfect when I call to other extensions on my local asterisk (1.2.0), but when I place calls to users on the remote box (boxes are linked via IAX2) audio

RE: [Asterisk-Users] Re: ip phone

Yep - I have one in my junk box. Maybe the SPA-841 would be a better choice for a few dollars more (haven't played with one personally, but everything I've heard says that they are much better than the GS BT's.) Having used both phones, I'd go the other way. The GS BT is a significant

RE: [Asterisk-Users] ip phone

looking for ip phones for an office setting. The client wants about 15 phones initially. Not counting volume discounts, does anyone have any recommendations. Cost is a factor, after discounts they were thinking about $50/phone. I've no idea what prices are like on the GXP-2000 on

RE: [Asterisk-Users] Bristuff / Junghanns / Customer Service

I bought a quadbri card from junghanns around two years ago. I've never dealt with the company in question, but isn't it a bit much to expect any company to take a product back after two years of use? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from

RE: [Asterisk-Users] g729 status in New Zealand

The question is really un-important for this list, it is ONLY important between the person who thinks that they can use the g729 codec ignoring the patent or considering that it is not legally enforcable for them and their lawyer who will give them concise information about the legal

[Asterisk-Users] Aastra 9133i registration errors

Hi all, I have a pair of new Aastra 9133i phones here that I'm testing for receptionist duty at a couple of places next week and they don't seem to be registering with * correctly. I've set the phone up with the following entries in the appropriate tftp config file: sip line1 auth name: 205 sip

[Asterisk-Users] TDM400 cards and modem/fax devices

Hi all, Having read the various fax and asterisk pages on voip-info, am I right in thinking I should be able to bridge Zap channels carrying fax without reliability problems (which as I understand things plague Fax-over-IP)? The reason for asking is in relation to a requirement where both fax

[Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

Hello all, I'm contemplating upgrading a client's asterisk system from 1.0.9 to 1.2 beta to take advantage of some of the new echo cancellers in the later zaptel packages. Problem is, I'll be doing it without physical access to the box and without being able to personally test the new echo

RE: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

Before you upgrade to 1.2 and potentially break a lot of things, have you followed the instructions available at http://www.voip-info.org/ wiki/view/Asterisk+zapata+gain+adjustment to adjust the rxgain and txgain? Don't suppose anyone knows of a 1004 Hz 0dB number I can call to test with

RE: [Asterisk-Users] Upgrading 1.0.9 to 1.2 beta

Incrementally reduce those gains by 2db per day and listen to your customer's feedback relative to echo. Don't bother using milliwatt generators and ztmonitor. (Those tools are okay to find a starting point if you have no other transmission test sets, but will not help even one little

[Asterisk-Users] Result branching in AEL

Morning all, I'm trying to rewrite my dialplan macros into AEL. How does one handle result-dependent branching (e.g. VoiceMail will branch to n+101 if mailbox doesn't exist) in AEL? Or is there a better way of doing this? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur

RE: [Asterisk-Users] Receptionist phones

I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_d ata_sheet09186a008008883d.html What are people using for receptionist phones that show all the

[Asterisk-Users] Problem dialling multiple SIP devices

I posted to the list with this issue a few weeks ago, but nothing really came of it. Either I'm missing something obvious (for which I apologize in advance) or this is a pretty serious issue between Asterisk and the SIP devices connected to it. I have 12 SIP phones at a particular site all

RE: [Asterisk-Users] Problem dialling multiple SIP devices

Drop the incoming calls into a call queue. Is it not the case that in order for calls to go into a queue, they must be answered first? Is it possible to drop calls into a queue before they're answered (by asterisk)? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is

RE: [Asterisk-Users] ATA-488 FXO

Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk to their Asterisk box (via SIP, of course)? I tried 2 of them at a client's site here in the UK. Is it possible to have such a beast operate reasonably? I was unsuccessful. The device would answer the line quite happily

[Asterisk-Users] Does AEL support arrays?

Hello all, Does anyone know whether there's any support in AEL for arrays, and if so, how one would go about implementing a shift statement? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (01604) 808408 Mobile: (07811) 332969 Skype: minotaur-uk

[Asterisk-Users] SIP phones supporting early dial

Hello all, Is there a list of phones that reliably support SIP early dial? One of the really nice things I've noticed about the 7960 (SCCP) is that each digit is sent straight to asterisk, so when the number has been completed, connection is almost instantaneous. I've tried early dial on both the

RE: [Asterisk-Users] TDM0xB vs. SIP for FXO

I am planning to connect my Asterisk PBX to one or two POTS lines, and am wondering if it is better to use a TDM card for this, or one or two SIP devices with FXO ports on them (such as an SPA-3000, Grandstream 488). I think it largely depends on where you're located and how much work has

[Asterisk-Users] Multiple zaphfc cards (for ISDN BRI) in a single machine

Hello all, I've just returned from a visit to a client site where their existing incoming lines are in the form of 5 ISDN BRI connections (for 10 channels total). We have successfully deployed Asterisk boxes with 2 HFC-based cards in the past, but I've no idea how well a standard PC will handle

RE: [Asterisk-Users] Multiple zaphfc cards (for ISDN BRI) in a singlemachine

I would suggest using a pair of 4-port cards. The interrupts alone from 5 PCI cards would kill most boxes. There is also an octo-card, but I have no personal experience of that. Hmm... the price is something of an obstacle - given that single BRI cards can be had for sub-£20, justifying £425

RE: [Asterisk-Users] TDM0xB vs. SIP for FXO

This is a very interesting thread. Could folks posting their experiences please also post the country their experiences relate to? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___

RE: [Asterisk-Users] SCCP support is making good progress

This is a old firmware issue, upgrading the phone firmware everything is working ok with the 7960 Sadly, that's the problem at the moment - I can't seem to get hold of new firmware for love nor money. Even the hunting for firmware on ebay route yielded zero results when I had a look yesterday.

RE: [Asterisk-Users] Dial with 44 and +44 prefix

One further question, how can I set up a line so that if 440 is dialled before a number the 0 is taken out so only 44 is actually used? exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3}) You could probably do it by playing around with different offets as well: exten =

RE: [Asterisk-Users] Dial with 44 and +44 prefix

To bad that prefixes like +220 (Gambia), +230 (Mauritius), +240 (Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 (Saint Helena), +350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 (Timor Leste), +680 (Palau),

RE: [Asterisk-Users] Dial with 44 and +44 prefix

I have inserted the lines you suggested but Asterisk keeps the 0 when dialling with all alternatives. Also, I am unsure what the phrase ${EXTEN::2}${EXTEN:3} does? Could you explain this abit? The syntax is {EXTEN:initial offset:length} So EXTEN:3 chops off the first three digits and

RE: [Asterisk-Users] SCCP support is making good progress

whoever owns a Cisco phone and is unhappy about slow firmware, incomplete XML support etc... should really have a look at Sergio Chersovani's rewrite of chan-sccp! Is there a good resource out there for people who don't have a lot of experience with Cisco phones? I picked up a 7960 earlier

RE: [Asterisk-Users] SCCP support is making good progress

Getting chan_sccp from Sergio to work is really easy. The distro contains a well documented sample config and - as I wrote before - there are lots of info in the chan-sccp-users mailing list archive. Yep, I've just tried chan_sccp with the 7960 I have here and it appears to work fine

RE: [Asterisk-Users] Grandstream GXP-2000

Erick, we're also using 1.0.1.12, having some echo problems, mostly with in/out going ZAP calls (on quadBRI, w/asterisk 1.0.9), the internal SIP calls seem to work fine. (but you have to make sure your volume isn't too high) Also the GXP-2000 has the annoying feature that calls get

[Asterisk-Users] Asterisk GUI/web interfaces that don't change config files

Hello all, I'm trying to find an Asterisk web interface (or windows gui interface) to asterisk that won't allow users to go making changes to config files. I've trawled through the very extensive list in the wiki, but there doesn't seem to be a clear defining line between applications that are

RE: [Asterisk-Users] ADSL

Is there an estimate on how many calls a 2Mb ADSL line can handle at the same time? Bearing in mind that the upload speed is 256Kb. Well, on our clients' ADSL connections (256k up and down) we seem to be able to push between 9 and 12 calls over it with g729 or gsm and iax trunking. Unless

RE: [Asterisk-Users] Dial with 44 and +44 prefix

exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1}) exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2}) How can I configure my extensions.conf to dial a number starting with 44 to dial without changes? Also a number sent from Outlook starting with +44? exten =

RE: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk?

So the idea is to put a SIM card inside the Asterisk box, equipped with a special card, a card which would be a mobile phone really. There are a number of places that sell GSM gateways (which is what you're referring to). What I've yet to see are GSM gateways for small business users that

[Asterisk-Users] Cisco 7960G and Asterisk

Hello all, I'm about to source a pair of 7960Gs to test with Asterisk prior to a demo to a new client next month. I've never used Cisco phones, let alone tried to make them play nice withly with *. According to our supplier, they either come with a SIP licence or a CCM licence (which from what

[Asterisk-Users] Delayed ringing on some SIP phones

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Wouldn't something like this work for you? [incoming-bri-one] exten = s,1,SetCIDName(Company 1) exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 1's 2's phones exten = s,4,Voicemail(su200) [incoming-bri-two]

[Asterisk-Users] Incoming call problem - ringing SIP devices report busy

bump from last week Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Why dont you make a special extension where you could provide the delay and the numbers you want to dial? exten = _900X,1,Wait(${EXTEN:4:2}) exten = _900X,2,Dial(SIP/${EXTEN:5}) then in the incoming context you could dial exten = s,1,Dial(SIP/200SIP/201LOCAL/90015300LOCAL/90015301)

[Asterisk-Users] Incoming call problem - ringing SIP devices report busy

Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so:

RE: [Asterisk-Users] country code list

441xxx geographic based landline 442xxx geographic based landline 443xx reserved 444xx reserved 445xx corp and voip 446xx reserved 447xx pagers, personal etc 448xx national rate, local rate, freephone, some mobile, blah 449xx premium services For the UK, your most accurate source of

RE: [Asterisk-Users] country code list

AstBill the Web-based open source Billing and Management software for Asterisk includes the information you are requesting. big snip Apologies for the slight threadjack, but as someone fairly new to the list, what *is* the policy on list advertising? There are quite a few posts I've seen in

[Asterisk-Users] Billing: amaflags and accountcode

Hi all, I have about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX-PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=,

[Asterisk-Users] Unwieldy outbound macro

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro

RE: [Asterisk-Users] Best drivers for HFC-S ISDN cards

I said: I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). Many thanks to those who replied. General consensus seems to be switching to mISDN or CAPI won't solve the intermittent echo

[Asterisk-Users] Best drivers for HFC-S ISDN cards

It seems that HFC-S cards can be connected with asterisk in a few different ways - isdn4linux, mISDN, chan_capi or zaphfc. I've tried isdn4linux (severe echo, reproducable on every inbound call) and zaphfc (intermittent echo, disappears within about 30 secs of the call starting). What's the

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