similar to: Problem faxing with SPA2100 in passthru mode.

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't.

Hi all, I hava SER with many clients (sipura SPA2100). One of these is an Asterisk which have others clients (sipuraSPA2100). I also have a Cisco GW which give me access to the PSTN. I make calls to all IP phones in my network, but I can't call PSTN numbers. After I dial, I hear 2 ringbacks but at the same time Asterisk says: Called pstn_number@SER_ip_address SIP/SER_ip_address-ec75 is

We're using STABLE CVS-Nv1-0-5-02/24/05 and we've been noticing that sometimes there's a socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs. If I issue a "sip show channels", I see thousands of "zombie channels". If I look into the details, that's what I get - actually one single "sip show channel

Hello All, I have been using the following phones with excellent success inside my lan: Cisco 7960 Polycom IP600 I have also been comfortable with the SPA-3000. I recently got a SPA-841 and the quality is aweful. Even in stock setup, it's like when I speak into the handset I sound like I am very far away. I want to get a hardphone for europe, but I need to be able to use a STUN server, and

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim.

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

Hello all ... mY first ever post in here. I am bit or (full) confused on what this program does.is it useful if i have a alcatel pabx system.And i can bill my guests for their call charges etc.. can i use it on calling another computer on the network via Ethernet card.Ihave already read the Documentation,But if any one could clear me up on the above things. how can i call a regular PSTN landline

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

I have two R objects, allDataSubset1 and allDataSubset2 and the str of both of them is shown below ( I don't show all 18 lists for space purposes ). The difference between them is that the times ( and possibly the days ) and the data is different and what I want to do is merge them so that only the data in Subset2 that is the same day and times as Subset 1 remains in the merged dataset.

My mistake sir. I was literally engrossed in my stupid logic, and while doing so, overlooked the simple and very effective solution you had offered. Sorry once again sir and will certainly try to be very careful in future. Thanks again and have a great weekend sir. Regards Amelia --- On Fri, 7/1/11, Petr PIKAL <petr.pikal@precheza.cz> wrote: From: Petr PIKAL