CRM Solution for Asterisk

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What is the best CRM solution for Asterisk, which is easy to deploy and Open Source? Well, there are some good options out there but the reality is that It's not possible to determine which one is "better". Nevertheless you can always evaluate and consider which one fits your needs, that's why among the different CRM solutions around, I would like to call your attention to Zurmo CRM, which have just released their 1.0 GA version. The first thing about Zurmo is that the way they care about the community's feedback, contributions and suggestions leads me to believe…

General 2.7 years ago 0 Answers

BT Fibre and 2701HGV

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Hi Does anyone have any experience of connecting SIP phones to an asterisk
server through the 2701HGV router that BT supply with their Infinity
product? Thanks in Advance Ish

Asterisk Users 3.2 years ago 2 Answers

One-way audio when calling multiple SIP

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Hi, On one of our locations, I am having issues with one-way audio when I call
several phones with SIP/Phone_A&SIP/Phone_B&SIP/Phone_C. When I call the
phones individually, they work fine, so it's not a volume setting on the
phone. Also this setup has worked at other locations. Any idea's what to look for? Thanks in advance.

Asterisk Users 3.2 years ago 0 Answers

911 multple-alert question

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Can you set up asterisk so when a 911 call is placed, in addition to the
call out to the PSAP, it also alerts multiple other phones on the switch
and will display detailed information. Such as alerting a receptionist
or security guard there is a 911 call elsewhere in the building and the
location of that call within the building? If so, how? Thanks in advance for the help...

Asterisk Users 3.3 years ago 6 Answers

Unclosed channel

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Dears, My scenario is to accept the call from user àAnswer the call -àplay mohà
dial(SIP/Trunk,XXXXX) The problem is when the user send the bye the trunk call will not hangup How to solve this issue exten => 446696,1,Ringing exten => 446696,n,Answer() exten => 446696,n,Wait(2) exten => 446696,n,Playback(Welcome) exten => 446696,n,Dial(SIP/Trunk/${EXTEN},300) exten => 446696,n,Hangup How to solve such issue Thanks in advance default iconwinmail.dat

Asterisk Users 3.3 years ago 1 Answer

Asterisk pickup call on first ring

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Hello,
Currently my asterisk system pickup incoming call after 3 or 4 rings.
How can I ask it to answer the call on the first ring? I put
immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no
different. Thanks in advance :) BR,
Anam

Asterisk Users 3.3 years ago 7 Answers

Common/Reasonable Assumption on DID/Channel over-subscription

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Hello All, just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX. I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have…

Asterisk Users 3.3 years ago 1 Answer

Linksys PAPT2

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Hello people, We have 4 asterisk server acting in which 2 are running as gateway, the
problem that i am facing is not asterisk related, we are using linksys PAP2T firmware 5.1.6 & 5.1.8, as gateway to some GSM
providers in Africa, we have now reached the point where we must put
credentioal into asterisk directly rather than using ATA with analoge
cards. hence the QTY of ATA and our needs are growing. We have every possible available soluation to find the SIP passwords inside
linksys PAP2T without joy, we have used various asterisk-password

Asterisk Users 3.3 years ago 0 Answers

Function not Registered??

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Hi all, I am running the same Asterisk 1.4.21.2 with the same configuration on all the servers in the region. I got this function called func_devstate which I use to control the lights of the Polycom phones. This module works well for all the Asterisk servers except this one. To get it to work, I basically compile this module together with the others and there is no need to explicitly load it in modules.conf. The problem is when my script uses function DEVSTATE, the Asterisk console shows that it is not registered. However, when I did a module show, it…

Asterisk Users 3.3 years ago 1 Answer